Showing papers on "Sampling (signal processing) published in 1972"

Computation of type curves for electromagnetic depth sounding with a horizontal transmitting coil by means of a digital linear filter

Abstract: The computation method described in this paper is based on the existence of a linear relationship between the mutual coupling ratio and the kernel function in the integral expression for it. Accordingly, the mutual coupling ratio can be determined by first computing sample values of the kernel function and then subjecting these to a digital linear filter. In the present paper the appropriate sampling distance is determined and the values of the digital filter coefficients are computed, both for electromagnetic sounding with horizontal coils and for electromagnetic sounding with perpendicular coils.

Broad-band thin-film signal sampler

Abstract: Previous developments in high frequency signal sampling have been extended by applying thin-film and solid state techniques to the sampler design. The basic elements of a sampler are a signal-carrying transmission line, a sampling gate, a drive line for the gate pulse, and a video output. In the design reported here, a thin quartz substrate supports the circuits. To reduce signal-line reflections the gate is made the center element of a low-pass signal-line filter. The gate pulse is developed across a length of slot transmission line in the signal ground plane parallel to and beneath the signal transmission line. A beam-lead Schottky diode pair is mounted across the slot with its center node connected to the signal transmission line through the substrate.

Multilevel signal transmission system

Abstract: Apparatus is disclosed for transmitting a multilevel signal together with at least one pilot signal of specified frequency as a timing signal for determining the sampling position of a transmitted signal and/or a signal for reproducing a demodulating carrier. Further, a reference level signal of a predetermined level is inserted in the multilevel signal train and the pilot signals are inserted in the multilevel signal train after frequency components in the neighborhood of the specified frequencies of the pilot signals are removed from the multilevel signal train on the transmitted side of the transmission line. In processing the multilevel signals the deviation of a sampled transmitted level of the reference level signal from the predetermined level thereof is detected; next, the frequency components in the vicinity of the pilot signals of the specified frequencies removed on the transmitting side of the line are extracted from the deviation; and then the signal distortion of the multilevel signal is corrected with the extracted frequency components on the receiving side on the transmission line.

Analysis, Optimization, and Sensitivity Study of Differential PCM Systems Operating on Noisy Communication Channels

Abstract: A closed-form analytical expression is obtained for the output signal-to-noise ratio (SNR 0 ) of differential PCM systems operating on noisy digital channels. A procedure is then proposed for maximizing SNR 0 by joint optimization of the quantizer, predictor, and sampling rate. Several examples are considered, including one- and two-sample feedback systems excited by speech and video (Gaussian) signals. The predictor used in the DPCM system is linear and time invariant, but otherwise arbitrary. As with PCM, contributions to output signal distortion can be expressed as a sum of three separate terms resulting, respectively, from quantization errors, channel transmission errors, and mutual errors arising from interaction between quantization and channel errors. When a conventional optimum linear predictor is used, transmission errors are shown to be no more serious for DPCM than for PCM. The results show that SNR 0 for a well-designed DPCM system is considerably higher than for a well-designed PCM system operating on the same digital channel, even if the channel is noisy. In considering examples, particular attention is devoted to determining maximum values for SNR 0 and to examining the dependence of SNR 0 on predictor coefficient values, number of quantization levels, sampling rate, message statistics, and channel error probability. Implications of this dependence on system design are noted.

Digital filters for sample-rate reduction

Abstract: The design of bandwidth-limiting filters for the purpose of sample-rate reduction is considered. Realization of linear-phase finite-duration impulse-response (FIR) filters for this application by direct convolution is shown to be more efficient than the recursive realization [1]. The degree to which the Nyquist rate (relative to the desired signal bandwidth) must be exceeded at the filter output in order to avoid aliasing is suggested as a measure of filter effectiveness. Direct convolution is faster than the fast convolution for FIR equiripple [2] filters designed to operate within 10 percent of the Nyquist rate with 60- to 70-dB stopband attenuation at a 2:1 sample-rate reduction. This advantage improves with the log of the sample-rate reduction ratio. Several comparisons made with recursive realizations of elliptic filters give the advantage to direct convolutional realization of FIR filters for sampling within about 20 percent of the Nyquist rate at 60- to 70-dB attenuation. Elliptic filters become more efficient at higher complexities (of about eight poles and eight zeros). Two design techniques that exploit the reduced output sample rate in the design of FIR filters by direct convolution are suggested. The effects of quantization of FIR filter coefficients on the frequency response are considered and several examples illustrated.

Inspection equipment for detecting and extracting small portion included in pattern

Abstract: Inspection equipment is provided which may easily detect and extract bad spots or defects included in a pattern such as an IC or printed circuit. The inspection equipment comprises a video input device for deriving the video information of a pattern to be inspected, a device for converting the output signal of the video input device into a binary video signal and sampling the binary video signal, a two-dimension buffer memory for converting the output of the A-D converter and sampling device into a two-dimensionally arranged signal, and a processing device for extracting the bad spots in the pattern from the output of the two-dimensional image extracting device. The output of the inspection equipment may be delivered to a TV display.

Signal detection system

Abstract: A signal detection system comprising detecting means for detecting the differential between a first and a second sampled value of an electrical signal subjected to sampling, comparing means for comparing the output of the detecting means with a predetermined first threshold value, discriminating means for discriminating the polarity of the output of the detecting means, adding means controlled by the output of the discriminating means for adding a predetermined second threshold value to the first sampled value, and means controlled by the output of the comparing means for deriving the result of addition from the adding means. The signal detection system serves for reducing or removing noises involved in or superposed on an electrical signal and more particularly a bioelectrical signal such as one recorded on an electroencephalogram or electrocardiogram.

Reduced rate sampling process in pulse code modulation of analog signals

Abstract: A composite color TV signal having the chrominance modulated and interleaved at the null points of the luminance frequency spectrum is comb filtered only over the region where the chrominance occurs. The filter separates luminance from chrominance. The chrominance is further demodulated and separated into its I and Q components. Each of the Y, I and Q components being a signal having a frequency spectrum with peaks at the horizontal line rate and nulls at an odd multiple of the horizontal line rate is sampled at a respective sampling frequency which is an odd multiple of half the line rate and which is also less than the Nyquist rate. Sampling at less than the Nyquist rate results in sampling energy being ''''folded back'''' into the spectrums of the respective Y, I and Q components; however, the sampling energy will be interleaved at the null points of the respective spectrums. When the sampled signals are received, the unwanted sampling energy is comb filtered out. All of the comb filters used comb only so much of the signal frequency band which includes the energy to be eliminated. Each comb filter includes a band pass filter which has a transfer characteristic which rises rapidly to a maximum level and remains constant over the frequency band of interest. The band pass filter includes phase equalization means which provides a linear phase shift versus frequency characteristic that intersects the ordinate of a phase shift versus frequency graph at a phase shift of 2n pi where n may be any integer.

Signal identification system

Abstract: A system is disclosed for reducing analog signal patterns or sequences to concise digital representations for identification, as to recognize an audio signal, for example, manifesting a musical recording. In the system, audio (analog) signals are sampled for dissection into a plurality of values which are accumulated over a sampling interval to provide an aggregate value. Preliminary to such a sampling operation, the signal of interest may be frequency dissected into a plurality of individual signals which are sampled then summarized or totalled. The aggregate total numerical values are reduced to a representative form by encoding them in accordance with their relative significance. Operation over repeated sampling intervals affords an expanded basis for indentification and recognition.

Digital phase shifter/synchronizer and method of shifting

Abstract: An electronic synchronizer for snychronizing the output pulse rate of an electronic clock with an input pulse train. The synchronizer provides a sampling signal output at a desired time within the time period of a single pulse of the input pulse train. The synchronizer phase-shifts the sampling signal by one-half of 1 clock cycle to either slow or speed the sample time, when required. A method of synchronizing by phase shifting the sample signal by one-half of 1 clock cycle is also disclosed.

Discrete adaptive delta modulation system

Abstract: In a discrete adaptive delta modulation system, the modulator located at the transmitter converts an analog signal into a digital signal at the rate fs while the demodulator located at the receiver converts the digital signal back into the analog signal. The modulator and demodulator each comprise a programmable pulse generator operating at the rate ft for providing a controlled number of pulses k during each sampling period 1/fs to its associated single stepsize analog integrator. The modulator further comprises a comparator, a quantizer, and a sampling pulse generator operating at the rate fs while the demodulator further comprises a low-pass filter in series with the integrator. The number of pulses k provided by the programmable pulse generator multiplied by the integrator basic stepsize sigma o determines the overall stepsize sigma k in the integrator output signal, where k ft/fs. A feature of this system is that the number n of available stepsizes sigma k, which is a function of the ratio of generator rates ft and fs, can be several hundred without affecting the complexity of the integrating circuitry.

Digital encoding system

Abstract: Methods of encoding in digital form an analogue signal including a modulated sub-carrier wave, such as a colour television video signal, are described in which the rate of sampling of the analogue signal is a simple factor, for example 3, times the frequency of the sub-carrier wave. The methods of encoding include comparing the instantaneous value of the analogue signal with a previously occurring value, which conveniently is spaced by one or more cycles of the sub-carrier from the instantaneous value, and then encoding the difference between the two values. From an N.T.S.C. signal the spacing between the two values may be one cycle of the sub-carrier wave, or about one line of the scan, (the actual spacing being an integral number of cycles of the sub-carrier wave, or both differences can be combined to produce a diagonal difference signal. The same spacings can be used for a PAL signal if a PAL modifier is used, otherwise a vertical spacing of two lines is necessary because of the change of phase of the sub-carrier in alternate lines. The differences from several spacings can be combined trigonometrically to synthesise a difference corresponding to a desired spacing, such as one cycle of the sub-carrier wave.

Priority channel scanning system with dual response time control

Abstract: A channel scanning and priority channel monitoring system for a multi-channel receiver includes a high frequency clock and a sequencing switch to rapidly scan the channels for signals and a low frequency clock and sampling switch to periodically monitor the priority channel for short time intervals during the reception of a non-priority signal. A variable response time squelch circuit is used to control the operation of the clocks. The squelch circuit operates in a fast response mode to rapidly sense the presence of a signal. After a signal has been acquired, the response time of the squelch circuit is determined by the strength of the signal being received, the squelch circuit operating in conjunction with a delay circuit to prevent the resumption of scanning during signal fades. A noise generator controlled by the variable response time squelch controls an audio muting squelch circuit in the receiver.

Delta modulator having low-level random noise characteristic

Abstract: A delta modulator for analog-digital conversion, comprising a high gain comparator amplifier connected to the input of a clock-controlled sampling flip-flop circuit; an audio signal is applied to the input of the comparator and a principal integrating circuit of short time constant develops a replica of the audio signal from the output of the flip-flop, and applies the replica to the comparator in bucking relation to the original audio signal. A self-bias variable D.C. reference signal is derived from the flip-flop output and is applied to the other comparator input; the bias circuit also transmits a negative feedback A.C. signal of limited amplitude, asymmetrical with respect to polarity, introducing a limited amplitude broad-band random noise into the demodulated output signal for idling (zero or very small input) conditions.

A real-time processing system and method utilizing discrete fourier transform

Abstract: A method of and device for carrying out real-time processing of electrical signals wherein, after sampling and quantizing the signals which are to be processed, real samples N are subjected to a pre-processing operation in a system in which a time sequence of said N samples of the processed signal is transformed into a sequence of N/2 complex samples Um applied to and processed in a F.F.T. iterative or repetitive algorithm computer unit of conventional design and mode of operation with N/2 points. The computer unit generates D.F.T. coefficients for which there is a symmetrical relationship between the even complex coefficients C2q and the related odd complex coefficients C*N 2q 2p 1.

Extracting circuit for reproducing carrier signals from a multiphase modulated signal

Abstract: A carrier reproducing circuit for demodulating a signal transmitted by a pulse coded modulation-multiphase modulation system in which the output of a carrier extracting circuit is sampled by a clock pulse of a bit repetitive frequency in a sampling circuit and the sampled output is applied to a band-pass filter having a pass band of a carrier frequency to reproduce a carrier of reduced phase jitter.

Synchronization method and an arrangement for recovery of binary signals

Abstract: A synchronization method and arrangement for recovery, at the receiver side of an information transmission equipment, of bit timing information during the transmission of a binary signal which at the transmitter side of the equipment is converted into a multilevel signal with correlative properties. From such multilevel signal, a binary signal conforming with the original binary signal is reconstructed at the receiver side. Bit timing information for sampling purposes at the receiver side is obtained by detection of the times at which the multilevel signal reaches and/or leaves at least one specific level.

Echo-sounding apparatus using fourier transformation

Abstract: An echo sounding apparatus having a source of coherent waves and an array of detectors each providing an output signal significant of the phase and amplitude of that portion of a wave reflected from the target to which the coherent wave from the source is directed. Sampling circuitry samples the outputs of the detectors at periods delayed with respect to a sampling origin time by an amount proportional to the square of the distance from the sampled detector to a reference point in the array. A calculator is then provided for carrying out a Fourier transformation on each sample to provide an output signal representative of the target.

Instantaneous adaptative delta modulation system

Abstract: An instantaneous adaptive delta modulation system including a modulator and a demodulator. The modulator comprises a comparator having a first input connected to receive an analog signal, a second input adapted for connection to an integrator located in the conventional feedback loop of the delta modulator, and a shift register connected to the output of the comparator and adapted to store the digits generated at the output of the comparator at sampling time T0 and the digits generated at sampling times T 1 and T 2. A binary up and down counter is connected to the shift register through a control circuit which makes the decision as to whether an up count, a down count, or no count at all is required. The output of the counter is applied to a decoder for converting the binary outputs of the up and down counter into a number of outputs. An amplifier having a corresponding number of values of gain is connected to the outputs of the decoder and is thus responsive to the level of the up and down counter for providing a predetermined gain into the feedback loop of the delta modulator. The integrator is connected to the output of the amplifier and its output is connected to the comparator which compares the output signal of the integrator with the input analog signal and generates a signal depending upon the difference between the two signals. The demodulator is similar to the modulator except that the input signal is fed directly to the shift register and that the output of the integrator is fed to a low-pass filter.

Electric field measuring and display system

Abstract: An apparatus for monitoring the electric fields of cloud formations within a particular area which utilizes capacitor plates that are alternately shielded from the clouds for generating an alternating signal corresponding to the intensity of the electric field of the clouds. A synchronizing signal is produced for controlling sampling of the alternating signal. Such samplings are fed through a filter and converted by an analogue to digital converter into digital form and subsequently fed to a transmitter for transmission to the control station for recording.

Voice synthesizer with digitally stored data which has a non-linear relationship to the original input data

Abstract: An electronic speaking machine has its vocabulary stored in a solid state memory so that the device, with the possible exception of the sound generator, employs no moving parts. The machine is capable of reproducing any spoken word by storing a digital representation of that word in its vocabulary. To reduce storage space, data compression is employed to reduce the data obtained from sampling an audio signal of the spoken word. Because only fixed words are stored, the data compression technique employed can be optimized for each stored word. A particular word is selected by applying the proper ''''select code'''' to the input of the apparatus. A ''''start of word'''' signal then causes a clock to sequence a counter through the addresses in the memory where the digital data representing the word is stored. Inasmuch as the stored digital data has a non-linear relationship to the original data, the non-linear data read out of the memory is transformed by a non-linear mapper to digital data having a linear relationship to the original data. A digital to analog converter transforms the linear digital values into an audio signal that is then filtered to obtain a reconstruction of the original audio signal of the spoken word. The reconstructed audio signal can then be used as the input to a conventional amplifier and speaker system.

Digital transmission system

Abstract: In the digital transmission system of the type in which from the transmitter are transmitted VSB- or SSB-modulated pulses of the sampling-point-zero-cross type multi-level pulse signal having the clock or pilot signal for timing and the carrier inserted at the ends of the band of said multi-level pulse signal, and at the receiving end said clock signal and said carrier are recovered to demodulate the modulated multi-level pulse signal, the intersymbol interferences at the sampling points spaced by one and two time slots apart from the center sampling point of the single pulse response in the transmission line are detected and used to automatically control the optimum phases of the recovered clock signal and carrier.

System for detecting dropout and noise characteristics of magnetic tape with switch means to select which characteristics to be detected

Abstract: Data bits represented by magnetic flux reversals at substantially each one-half cycle of uniform data periods are recorded on each of a plurality of data channels of the magnetic tape which is to be tested for dropout and noise characteristics. The recorded data bits on each of the plurality of data channels are then detected by a suitable means such as a read-write head. The detected data bits for each channel are connected as the input to a bistable means which in turn provides one of the inputs to an associated sampling gate employed for the purpose of detecting each dropout of a data bit for each channel. All channels are connected in common to an appropriate means for generating a timing signal in response to the first received output signal from the detected data bits. The timing signal is used to develop a first delay of less than one-half of the data period of the originally recorded data and applied to the sampling gates as their second input, actuating each noninhibited sampling gate to produce a dropout error signal. The timing signal also is used to develop a second delayed signal which is connected to reset the bistable means after the sampling has been completed. Digital counter means are connected to receive and count the dropout error signals thus developed by all the channels. In a preferred embodiment an appropriate chart recorder may also be connected to receive and record the dropout error signals relative to time distribution. Additionally, a digital counter and chart recorder may be connected in each channel to count dropout error signals within each channel and also record error time distribution over the length of tape being tested.

Apparatus and method for the measurement of acoustic absorption and impedance

Abstract: Apparatus and method for measuring the acoustical impedance and/or the absorption coefficient of materials of the class which exhibit nonlinear behavior at high sound intensities. A high-intensity biasing noise having a broadband spectrum and a low intensity pure sinusoidal tracer tone are introduced in combination into a standard impedance tube or standing-wave apparatus. The standing-wave pattern in the tube is measured by a pressure microphone probe movable along the axis of the tube. The tracer tone is retrieved by highly selective filtering and used to provide a measurement signal which corresponds to the response of the material. By sampling the standing wave pattern in the tube, the entire spectrum of the broadband noise response may be obtained.

Apparatus for sampling an asynchronous signal by a synchronous signal

Abstract: The asynchronous signal is complemented and directed to a latching logic circuit as the data input and the synchronous signal are directed to the clock input of the latching circuit. The latching circuit has built-in delays to reliably latch if the low or active portion of the asynchronous pulse occurs during the "window" or high portion of the synchronous signal. The latching circuit also includes a jamming circuit connected to its clock input whereby a low or disabled window time of the synchronous signal prevents a change in state by the latching circuit. A pulse delay circuit generates a sampling pulse a period of time after the window time to sample the output of the latching circuit.

Pattern discrimination system using television

Abstract: A pattern discrimination system for automatic and rapid detection of occurrences of an abnormal condition in a pattern under surveillance comprising the means and steps of producing at least one horizontal gate pulse and vertical gate pulse of any required width and position by utilizing the television art, sampling a video signal with the horizontal and vertical gate pulses to form at least one sample surface variable in position, size or shape thereof in the field of view of a monitor, producing an integrated value of voltage of a video signal corresponding to the sample surface of the pattern under surveillance, digitally storing the integrated value of voltage of the video signal in a digital memory, and producing another integrated value of voltage of the video signal corresponding to said sample surface of the pattern under surveillance after a lapse of a predetermined time interval and comparing the second integrated value of voltage with the integrated value of voltage stored previously.

Television multiplexing system

Abstract: A television multiplexing system which includes a circuit that inserts a digital coded sync signal and a digital code into a video signal for identifying the channel from which the video signal was generated so that a plurality of signals can be sent over a single hard-line The digital sync signal and the digital coded signals are generated by a single crystal controlled clock so that they are always in synchronism with each other In demultiplexing the signals so as to feed the video signal to a proper recording channel the sync signals are utilized for shifting the digital coded signals into a shift register and the shift register, in turn, activates a decoder according to the code stored in the shift register for selecting the proper recording disc or receiver for storing the video signal

Design and Simulation of Digital Filter Banks for Speech Analysis

Abstract: Short‐time spectrum analysis is the basis for many speech analysls systems. Although the fast Fourier transform is generally used to perform spectrum analysis on a general purpose computer, a bank of recursire digital bandpass filters may be the best approach for hardware realizations. This paper discusses the analysis and design of digital filter banks composed of equal‐bandwidth, equally spaced, bandpass filters. It is shown that significant improvement in the frequency response of the composite filter bank can be achieved by appropriate choice of the relative phases of the bandpass filters. Also discussed is an efficient general purpose computer simulation of a bank of recursire digital filters as required, for example, in a phase vocoder analyzer [Flanagan and Golden, Bell Syst. Tech. J. (Nov. 1966), This simulation uses the fast Fourier transform to compute filter outputs at a low sampling rate (approximately 100 Hz). For synthesis, the spectrum parameters are interpolated to a 10‐kHz sampling rate u...

Automatic presetting of vibrator phase-lock circuits

Abstract: In seismic geophysical surveying using one or more servocontrolled vibrators as the source of varying-frequency wave energy, phase-locking of the vibrator output to the input pilot wave form is essential to the proper compositing of both sequentially received waves and simultaneous inputs by two or more vibrators. By the present invention, the phase lock-in time during which non-coherent signals may be produced is minimized by automatically sampling the phase and the gain errors shortly following each signal initiation, and using the samples to update the preset values for each subsequent signal generation.

Analog to digital converter for electrical signals

Abstract: Unknown bipolar analog signals are converted to equivalent digital signals by comparison with positive and negative reference voltages. Sampling is performed for a preselected time period during which the reference voltages are alternately compared against the unknown analog signal. The reference polarity is switched each time the comparison result indicates a polarity change. Counting of clock pulses during all times one of the reference voltages is applied for the preselected period results in the digital equivalent of the analog input.