Showing papers on "Sampling (signal processing) published in 1982"

Wave propagation and sampling theory—Part I: Complex signal and scattering in multilayered media

Abstract: From experimental studies in digital processing of seismic reflection data, geophysicists know that a seismic signal does vary in amplitude, shape, frequency and phase, versus propagation time To enhance the resolution of the seismic reflection method, we must investigate these variations in more detail. We present quantitative results of theoretical studies on propagation of plane waves for normal incidence, through perfectly elastic multilayered media. As wavelet shapes, we use zero-phase cosine wavelets modulated by a Gaussian envelope and the corresponding complex wavelets. A finite set of such wavelets, for an appropriate sampling of the frequency domain, may be taken as the basic wavelets for a Gabor expansion of any signal or trace in a two-dimensional (2-D) domain (time and frequency). We can then compute the wave propagation using complex functions and thereby obtain quantitative results including energy and phase of the propagating signals. These results appear as complex 2-D functions of time and frequency, i.e., as “instantaneous frequency spectra. ’ ’ Choosing a constant sampling rate on the logarithmic scale in the frequency domain leads to an appropriate sampling method for phase preservation of the complex signals or traces. For this purpose, we developed a Gabor expansion involving basic wavelets with a constant time duration/mean period ratio. For layered media, as found in sedimentary basins,

Bandpass Signal Sampling and Coherent Detection

Abstract: Coherent detectors in radar and communications receivers are generally implemented in the form of two parallel baseband channels which form in-phase (I) and quadrature (Q) components of a received RF/IF signal. Phase errors of several degrees due to imperfect matching of these separate channels limit the performance achievable from signal processors such as moving target indicators (MTI), coherent integrators, Doppler filters, antenna array processors, and coherent sidelobe cancellers. Thus methods in which a single analog to digital (A/D) converter samples and digitizes the IF signal directly, eliminating the need for IF to baseband conversion, have been of recent interest and are the subject of this paper. To obtain accurate coherent detection from IF samples taken near the Nyquist rate requires interpolation based upon a number of stored samples. An algorithm derived from sampling theory is defined and used to demonstrate accurate reconstruction of the original IF signal from digitized samples. In-phase and quadrature components of the signal are shown to be available from processed samples with demonstrated phase errors less than 0.2°.

Signal sampling system

Abstract: A signal sampling system includes a combined digital and analog acquisition time base for accurately sampling and displaying an extremely short duration electrical event superimposed on a long time duration signal. The signal sampling system of the present invention is intended for use in stimulus-response situations, and both stimulus and response signals are synchronized with the clock of the digital portion of the time base. The system may be operated under microprocessor control, providing both flexibility and programmability, in turn permitting not only acquisition of waveforms that start and stop at arbitrary points with extreme precision, but signal averaging or smoothing as well.

Estimating the parameters of exponentially damped or undamped sinusoidal signals in noise

Abstract: : We address the problem of estimating the parameters of a signal embedded in noise The signal is composed of samples of a sum of M exponentially damped or undamped sinusoidal signals We suggest improvements to the original approach proposed by Prony He observed that the samples s(n) obey an M th order difference equation and that from the roots of the characteristic polynomial of the difference equation, the parameters (sk) can be determined We present three methods to improve this approach when the signal is noise corrupted

Spectrum surveillance receiver system

Abstract: A method and an apparatus for surveilling RF signals, the apparatus comprising a receiver for receiving RF signals at a selectively variable reception frequency and converting any signal received at that frequency to a predetermined IF signal, a sampling circuit for sampling the IF signal and providing output signals representative of a plurality of predetermined properties of the received signal; a circuit for comparing the representative output signals against a plurality of sets of reference signals each of which is representative of a particular modulation type and providing a first output indicative of the type of modulation of the received signal when the representative signals substantially match the set of reference values of one of the modulation types and a second output indicative of an unidentified received signal when the representative signals do not match any of the sets of reference signals.

Real time digital scan converter

Abstract: A digital scan converter is disclosed wherein signal information supplied by a sector scanning surveillance system relative to a polar coordinate system is converted to a signal for driving a television-type display or other Cartesian coordinate device by: (a) sampling the signal associated with each consecutive scanning path of the surveillance system at a rate determined by the azimuthal angle that defines the scanning path of interest; (b) storing each set of signal samples as a column of data in a rectangular memory array: (c) accessing the stored data on a row-by-row basis; and (d) utilizing a previously determined mapping strategy to cause each accessed signal sample to form a segment of a line of display within the Cartesian-formulated display devices so that the length of the segment formed by each signal sample is determined by the row and column address of the storage location that is associated therewith when forming a television compatible signal, each signal sample dictates video signal level during a portion of a corresponding horizontal sweep period that is determined by the mapping strategy.

CCD Demodulator circuit

Abstract: A semiconductor integrated circuit using charged coupled device (CCD) technology for performing demodulation of time-varying signals which have been phase or amplitude modulated. The CCD circuit performs a sampling of the time-varying signal at a suitable sampling frequency depending upon the frequency of the phase or amplitude modulation of the carrier. The CCD device converts the sample into an equivalent charge packet which is used to control the control electrode of a field effect transistor in an amplifier circuit. The magnitude of the sample is representative of the amplitude of the carrier so that the output of the field effect transistor represents a demodulated signal. The circuit is a broad spectrum device, operable with a signal frequency from the audio into the gigaHertz (GHz) frequency range.

Periodic wave elimination by negative feedback

Abstract: Means and method for substantially eliminating a selected periodic wave from a frequency band-limited combined wave containing other waves. The combined wave is sampled in a sampling gate, opened by short gating pulses with a frequency equal to the selected frequency divided by an integer, and greater than the Nyquist frequency for the combined wave, timed from the selected periodic waves. The samples are reconstructed in a filter with the pass-band equal to the frequency band of the combined wave. The direct-current component in the output of the sampling gate, due to any error in the timing of the instants, is used in a negative-feedback circuit to adjust the sampling instants so that the periodic wave is substantially completely eliminated.

Automatic background noise suppressor

Abstract: A circuit for suppressing background noise of a continuous nature while enhancing speech signals, or signals having the transient temporal qualities of speech, includes a signal multiplier which, in the preferred embodiment, receives the composite audio signal along with a control signal present only when the speech component of the audio signal is present. The control signal may be derived from an AGC circuit having a slow attack, fast decay characteristic to establish a quiescent output level from the AGC amplifier in the absence of speech. An envelope detector is biased to provide a zero output amplitude in response to the quiescent amplifier output level. Speech components appearing in the amplifier output signal are then envelope-detected and filtered to provide the control signal. Alternatively, the control signal can be derived by envelope-detecting the audio signal, filtering the detected signal to remove its d.c. component representing the continuous noise, and then detecting and filtering again. In still another embodiment, the control signal acts upon a constant amplitude instead of the audio input signal in order to provide a speech-responsive tactile vibration for the deaf.

Speech sampling and companding device

Abstract: An analog speech signal is sampled of a nominal rate of 6 kilohertz and digitized in a Mu-Law Encoder. The digital output of the Mu-Law Encoder is converted by a microprocessor performing table look-up to linearized pulse code modulation (PCM) samples nominally of eight bits per sample. Using a BSPCM (Block Scaled Pulse Code Modulation) method, in each block of nominally 246 eight-bit PCM samples (representing approximately 41 milliseconds), the maximum and minimum sample values are found and used to calculate a scale factor equal to the maximum sample value minus the minimum sample value, with the difference being then divided by a constant number nominally equaling 16. Then the BSPCM samples are generated from the PCM samples each as a corresponding one PCM sample minus the minimum PCM sample value, the difference being then divided by the scale factor. In effect, the bit rate is reduced by adjusting the step size to follow the local block dynamic range. The BSPCM samples so created are susceptible to signal processing operations like as to PCM samples. When the BSPCM encoded words plus the minimum, PCM encoded, sample plus the range increment scale factor are stored as a data block, then such data block can, at a later time, be decoded, or reconstituted, into linear PCM data. A silence interval is encoded as zero amplitude using run length coding of the number of blocks. Such digital PCM data can be converted to an analog audio signal for voice output across a telephone system.

Capacitively coupled machine tool safety system having a digital quantizer

Abstract: A machine tool safety system having a signal induced in a capacitive receptor antenna by the action of an electric field capacitively coupled thereto includes a receiver having an array of linear amplifiers with fast response times, a network connected to the output of each amplifier and responsive to a first command to sample the amplifier output signal, an analog-to-digital converter associated with the sampling network to quantize the signal from the first amplifier in the array that is not saturated, and a network responsive to a second command to dump the sampled signal value prior to the occurrence of the next sample time.

Digital Protection of Power Transformer Based on Weighted Least Square Algorithm

Abstract: The paper presents a new weighted least-square scheme for differential protection of power transformers using digital techniques. The differential current waveform is modeled as a sum of decaying de, fundamental and selected higher harmonic components. The model coefficients are established by using a least-square filtering technique for the sampled data. A root mean square error criterion is established to select the proper harmonic orders, data window, weighting matrix and sampling rate for fast detection of internal faults and high restraint against energizing transients in transformers. A variety of simulated waveforms is presented to highlight the important features of this algorithm. The results clearly demonstrate the efficacy and weakness of the weighted least square model.

Electromagnetic flow meter

Abstract: PURPOSE:To form a transmission path between a detecting portion and a converting portion into a single system being a flow signal transmission path, by transmitting a flow signal, which is overlapped with a synchronous signal serving as a trigger, from the detecting portion to the converting portion. CONSTITUTION:An exciting power source oscillating part 14 in a detecting part 10 is controlled by a timing signal generating part 16, and generates an exciting signal such as rectangular wave whose polarity reverses at intervals of a given time. In an overlapping part 17, the exciting signal is overlapped with a synchronous signal, and with a coil 2 excited, a flow signal is fetched from electrodes 13a and 13b of a conduit 11. The flow signal is transferred to a converting part 20 through a transmission path 30, and a converting flow signal, synchronizing with the detecting part 10 and being sampled, is outputted through a sampling gate 23 controlled by means of a sampling signal, a trigger of which an overlapping synchronous signal from a sampling signal generating part 22 serves as. As a result, a transmission path 30 between the detecting part 10 and the converting part 20 can be formed into a single system being a transmission path of a flow signal from the detecting part 10, and the converting part can be installed at a remote place without a trouble.

Method and apparatus for detecting tracking error

Abstract: A tracking error of a light spot focused on an optical record disc by means of an objective lens with respect to an information track recorded on the disc is detected by processing outputs from a light detector arranged in a far field of the track and having four light receiving regions on which a light reflected by the disc is incident. A tracking error signal of a differential detection system, i.e. a difference between two sums of outputs from two pairs of two regions aligned in a direction tangential to the track is derived, and a push-pull signal, i.e. a difference between two sums of outputs of two pairs of two light receiving regions aligned in a radial direction is also obtained. Zero crossings of the push-pull signal are detected to produce sampling pulses and the tracking error signal of differential system is sampled by the sampling pulses to derive a tracking error signal which is substantially free from disturbance due to a possible shift of the light impinging upon the light detector with respect to the optical axis of objective lens.

4‐bit 828‐megasample/s electro‐optic guided‐wave analog‐to‐digital converter

Abstract: Individual bit channels of a 4‐bit guided‐wave electro‐optic analog‐to‐digital converter have been demonstrated at 276 and 828 megasamples per second. The converter consists of a mode‐locked Nd: yttrium aluminum garnet (YAG) laser for sampling, a LiNbO3 Ti‐indiffused waveguide interferometric modulator array for conversion, a Ge avalanche photodiode (APD), and a special 1‐GHz Si integrated circuit for digital processing. Beat‐frequency tests with a 413‐MHz test signal, representing the highest frequency analog waveform converted, are reported.

A 2-channel, 16-bit digital sampling frequency converter for professional digital audio

Abstract: A digital sampling frequency converter for arbitrary ratios of sampling frequencies is presented. It is based on a multistage interpolating filter, and on a novel time-domain control of the filter stages by signals derived from the sampling frequency clocks. Time-domain resolution of ±300 picoseconds is obtained, compatible with digital audio of 16-bit resolution. In addition to the filter design and implementation, measurement results are presented. They indicate that 16-bit accuracy is indeed achieved, even with asynchronous, drifting and time-varying sampling frequencies. A number of applications (digital mastering, program transfer between conflicting digital audio formats, pitch control with constant sampling frequency in digital recorders, error concealment, interfaces in digital transmission) are presented.

Delta-Sigma modulator with switch capacitor implementation

Abstract: A Delta-Sigma ΔΣ modulator for one-bit analog to digital conversion of a signal, such as a telephone voice channel switch to digital form is disclosed. The circuit incorporates integration operative as low-pass filtering. The voice signal is sampled at a rate high compared to the highest voice channel frequency component. Sampled instantaneous signal values are integrated, re-sampled and again integrated to reduce encoding noise. Finally, a bi-level sense identifying circuit links the second integrator output to a D type flip-flop providing the one-bit encoded output. The sampling and integrator control is effected by switching in a capacitor charge shifting arrangement under clock control and feedback is applied as a capacitor switching program modifications.

FM/CW radar linearization network and method therefor

Abstract: An FM/CW radar linearization network provides target identification data discriminating a target from background reflections and/or false targets by compensating for random variations in the linearity of the frequency sweep of the radar transmitter in the processing of the radar receiver signal, the latter being characteristic of the range and physical size of the target. Linearization is achieved by sampling the transmitter signal, generating an error signal proportional to and indicative of any deviations in the frequency sweep of the transmitter from that for linear operation, and applying that error signal to correct the target data signal, thereby frequency normalizing the same for processing and utilization. A number of alternatives in respect of the development of an appropriate error signal and further alternatives in its use as the basis for data correction are disclosed, including sampling of the transmitter in respect of the phase angle over a predetermined period or the time for a predetermined number of cycles, in order to develop an error signal proportional to and indicative of any deviations in the frequency sweep of the transmitter from that were its operation linear; while data correction may be made in one of a number of alternative ways, including phase rotation of the raw target data input signal, time-shifting of the sampling rate of such data, or frequency mixing the raw target data with an error control signal. Representative networks and suitable methodologies to achieve linearization are disclosed herein.

An accurate digital instrument for the measurement of circular polarisation of luminescence

Abstract: An apparatus to detect circular polarisation of luminescence (CPL) is described. The instrument, involving a digital synchronous sampling photon counting system and an effective square-wave modulation, is absolute and has a high sensitivity. Its baseline is flat and stable, enable prolonged measurements to detect small degrees of circular polarisation, to study poorly fluorescing compounds or to detect circular polarisation in chemiluminescence. An additional advantage is that the instrument can be simply switched to measure linear polarisation.

Digital coherent psk demodulator and detector

Abstract: An improved digital demodulator and detector for phase-shift-keyed digital signals providing coherent detection with high sensitivity and stability on noisy channels. Digital sampling is used in conjunction with a digital phase locked loop system in which basic time increments are subtracted or added to an equilibrium timing loop to provide simultaneous carrier demodulation and bit recovery.

Multi-mode oscilloscope trigger with compensating trigger delay

Abstract: In a sampling digital oscilloscope means operable to measure the time period from the occurrence of a trigger signal to the onset of signal sampling period so that commulatively stored samples of a repetitively sampled signal visually coincide, producing a clear, jitter-free signal image. Moreover, the trigger signal is also produced according to a hold-off mode to allow the user to evaluate the results of processing the sampled signals by the digital oscilloscope before adding new data samples. In addition, a triggered update of stored data by new signals is provided, which is further controllable via an arm/disarm control to interrupt realtime updating of signals, allowing the user to observe signal changes as they occur.

Method and apparatus for checking analog-to-digital converters, digital-to-analog converters, or telecommunications transmission links which contain such converters

Abstract: The test specimen is charged with an analog, preferably sinusoidal periodic test signal or, respectively, with the digital samples of such a test signal, whereby a numerical relationship exists between the period of the test signal and the system-inherent sampling period such that all sampling times lying within a predetermined test time span occupy different relative time slots in the periodicity interval of the test signal. Arising as a result thereof are a plurality of output information of the test specimen which, as a totality, describe the response of the test specimen to the test signal as precisely as desired. Said output information are, if need be after analog-to-digital conversion in a standard coder, deposited in a store and are available for identifying the desired properties of the test specimen, for example, the level-dependent distortions, by means of a computer.

Presampling filtering, sampling and quantization effects on the digital matched filter performance

Abstract: Due to the increased capability and reduced cost of digital devices, there has recently been a growing trend to digitize the matched-filtering data detector in the receiver. Comparing with an idealized integrate-and-dump analog matched filter, the digital matched filter (DMF) requires more Eb /No in order to achieve the same bit error rate performance because of the presampling filtering, sampling, and quantization effects. This paper analyzes the performance degradation resulting, separately and jointly, from these three effects. Quantitative results are provided for commonly chosen sets of design parameters. For a given performance degradation budget and complexity limitation, these results could be applied to choose the optimum DMF design parameters including the presampling filter bandwidth, the sampling rate, the number of quantization bits, and the spacing between adjacent quantization levels. 1.0 INTRODUCTION The study of sampling and quantization effects on the digital matched filter (DMF) has recently received much attention in evaluating the performance of digital receivers that employ matched-filter detection.[1, 2, 3] A fundamental case of interest is the case when the input to the DMF consists of (1) an NRZ-L PCM baseband signal and (2) an additive white Gaussian noise process. The NRZ-L PCM signal appears in the time domain as a train of rectangular pulses of voltage levels +V or -V (see Figure 1), depending on whether the transmitted data bit is a 0 or a 1. For such a signal plus noise, it is well known that the integrate-and-dump filter is the optimum (or the matched-filtering) detector which results in the minimum error probability as shown in Figure 2. The increased stability, reliability, and flexibility, as well as the decreased size and cost make the digital implementations of many analog matched filters highly desirable. Figure 3 illustrates one possible digital implementation of the integrate-and-dump filter. As evident from the figure itself, the performance of this digital integrate-and-dump (matched) filter depends upon three system parameters: (1) B (Hz), the bandwidth of the presampling low-pass filter (2) fs (samples/bit), the sampling rate of the sampler in samples per data bit (3) m (bits), the number of bits of the quantizer Because of presampling filtering, sampling and quantization effects, the DMF requires more Eb/No than the analog matched filter. Thus, the degradation factor D of the DMF may be defined as the required increase in Eb/No for the DMF in order to yield the same error probability as the analog matched filter. In what follows, the degradation factor is derived in detail with quantatative results presented for commonly chosen sets of design parameters. 2.0 ANALYSIS This section is devoted to deriving the error probabilities and hence the degradations for the DMF. Refer to the block diagram of the DMF in Figure 3. Let the received signal plus noise at the input to the DMF be expressed as x(t) = s(t) + n(t) (1) where n(t) = a stationary white Gaussian noise process of two sided spectral density No/2 = a rectangular pulse train of voltage levels +V or -V and u(t) = a rectangular pulse of amplitude V ana duration T The energy per bit to one-sided noise density ratio is hence given by

Apparatus for testing circuits within a system

Abstract: A capability for internal self-testing is provided for a test instrument in which data, in digital and in analog form, is sampled from various test points. A programmable delay generator is used to process some of the data and to provide strobes which control the data output and which control the sampling of data. This enables measurements to be made at a preselected time after the occurrence of a particular signal, thereby giving the system a capability of using the delay generator not only for providing comparisons of time functions but also for initiating test sub-routines. The self-test capability enables test equipment having time-dependent functions, such as test equipment for video displays, to be verified with respect to accuracy in order to establish a confidence test.

Method and apparatus for detecting tracking error

Abstract: An optical disc player including an automatic tracking control for causing a light spot projected on a rotating optical disc to trace spiral or concentric information tracks is disclosed. A tracking error is detected by a wobbling method in which the light spot is vibrated in accordance with a position modulating signal having a given frequency, an amplitude modulated component due to a positional modulation is derived from a reproduced RF signal by passing an envelope signal of the reproduced signal through a band pass filter having a center frequency equal to the frequency of the position modulating signal, a sampling pulse is produced in synchronism with the position modulating signal and the amplitude modulated signal is sampled by the sampling pulse to derive a tracking error. A phase difference between the amplitude modulating signal and the sampling pulse is detected and a phase of the sampling pulse is modulated in accordance with the detected phase difference in such a manner that the amplitude modulated component can be always sampled at its peak points.

2-bit 1 Gsample/s electrooptic guided-wave analog-to-digital converter

Abstract: An electrooptic analog-to-digital converter which uses a pulsed GaAlAs diode laser for sampling, a LiNbO 3 Ti-indiffused-waveguide interferometric array for conversion, a Si APD, and a special 1 GHz Si integrated circuit for digital processing has been demonstrated at 1 Gsample/s Individual bit channels of this 2-bit device have been tested by a beat-frequency technique with a 4992 MHz test signal This represents the highest frequency conversion of an analog waveform reported to date

Digital coherent detector

Abstract: A digital coherent detector for sampling a band-limited IF signal directlyo obtain its in-phase and quadrature coefficients I(t) and Q(t) without using quadrature channels comprising: an A/D converter for sampling and digitizing an IF signal r(t)=I(t) cos 2π fo t-Q(t) sin 2π fo t at a sampling frequency of fs =2fo /(M-1/2)=2W. where M is an integer, W is the bandwidth of the input signal, and fo is the center frequency of the signal, so that the sin and cos sinosoidal terms alternately go to zero and one respectively, thereby alternately yielding the sample values I(t) and then Q(t); a finite impulse response digital filter for estimating the value r(t) for the coefficient whose sinosoidal term has gone to zero in a given sample with the function ##EQU1## where s(t-n/2W) is a self truncating interpolation function; and a switching circuit for properly setting the signs for r(tm) and r(˜tm) and switching these terms between I and Q output lines in accordance with a sampling clock pulse.

Multilevel signal compression method

Abstract: A digitized multilevel signal is encoded on the basis of unit combinations each including a run-length bit field indicative of run-length at each of the signal levels and a single continuation bit indicative of transition of the signal levels. The encoding is so modified that a virtual run of zero length is inserted when the transition of the signal levels at adjacent sampling points departs from a predetermined order. Polarity of the continuation bit is inverted upon every transition of the signal level.

Low frequency digital comb filter system

Abstract: A comb filter arrangement operating at a reduced data rate is provided, which requires comparably fewer storage locations than previous arrangements. A digitized composite video signal of a given codeword rate is applied to a first bandpass filter, which produces a filtered signal restricted to a portion of the passband of the composite video signal. The filtered signal is then subsampled at a rate which satisfies the Nyquist criterion for information of the restricted passband. Codewords, now at a reduced data rate, are applied to a one-H delay line, and delayed and undelayed signals are combined to produce a first comb-filtered signal. The first comb-filtered signal is then applied to a second bandpass filter, which provides a sequence of codewords at the codeword rate of the original digitized composite video signal over a given frequency band. This sequence of codewords is then combined with the codewords of the composite video signal to produce a second comb-filtered signal.