Showing papers on "Digital signal processing published in 1969"

A Fast Fourier Transform Algorithm Using Base 8 Iterations

Abstract: This collection of papers is the result of a desire to make available reprints of articles on digital signal processing for use in a graduate course offered at MIT. The primary objective was to present reprints in an easily accessible form. At the same time, it appeared that this collection might be useful for a wider audience, and consequently it was decided to reproduce the articles (originally published between 1965 and 1969) in book form.The literature in this area is extensive, as evidenced by the bibliography included at the end of this collection. The articles were selected and the introduction prepared by the editor in collaboration with Bernard Gold and Charles M. Rader.The collection of articles divides roughly into four major categories: z-transform theory and digital filter design, the effects of finite word length, the fast Fourier transform and spectral analysis, and hardware considerations in the implementation of digital filters.

Laser-Acoustic Signal Processing

Abstract: The variable delay and processing of RF and microwave signals is a current problem of considerable importance both for the many military applications in radar and electronic counter-measures and for the civilian applications in signal processing. The very low velocity of acoustic waves, as compared to the propagation velocity of electromagnetic signals, permits the fabrication of physically compact systems capable of storing and processing electromagnetic signals. Among the several suggestions for obtaining a continuously variable delay, the use of the laser-acoustic interaction has been shown to be among the most promising.

Digital filtering with sampled signal spectrum frequency shift

Abstract: A frequency shift of the sampled signal spectrum equal to half the sampling frequency is considered for digital filtering. The application to an efficient bandpass analysis is described. Some experimental results are reported.

Papers on digital signal processing

Abstract: A collection of articles originally intended for use in a graduate course. This collection of papers is the result of a desire to make available reprints of articles on digital signal processing for use in a graduate course offered at MIT. The primary objective was to present reprints in an easily accessible form. At the same time, it appeared that this collection might be useful for a wider audience, and consequently it was decided to reproduce the articles (originally published between 1965 and 1969) in book form. The literature in this area is extensive, as evidenced by the bibliography included at the end of this collection. The articles were selected and the introduction prepared by the editor in collaboration with Bernard Gold and Charles M. Rader. The collection of articles divides roughly into four major categories: z-transform theory and digital filter design, the effects of finite word length, the fast Fourier transform and spectral analysis, and hardware considerations in the implementation of digital filters.

Digital pulse stretcher

Abstract: A digital signal processing arrangement involving pulse forming and timing. A digital pulse stretcher synchronizes the output of an asynchronous pulse source with a synchronous digital frequency source. The pulse stretcher includes a first flip-flop which is set until the asynchronous pulse arrives. Arrival of the asynchronous pulse resets the first flip-flop and simultaneously sets a second flip-flop. A third flip-flop is set until the synchronous pulse arrives at which time the third flip-flop resets. As the end of the synchronous pulse the second flip-flop resets, causing the third flip-flop to return to the set state. A triggered output flip-flop is also included and operates to release the output of the second flip-flop at the beginning of the next bit time.

Analog/digital differential apparatus for comparing resolver output data with a digital signal

Abstract: Apparatus for comparing an analog input signal with a digital input signal and providing an analog output signal proportional to the difference therebetween, said apparatus comprising an analog/digital differential controller including digital-toanalog converters responsive to the more significant bits of the digital signal and operating in combination with multiplier and subtractor devices for establishing a coarse relationship between the analog and digital input signals and supplying an accurate reference voltage to a fine resolution digital-to-analog converter which is responsive to the less significant bits of the digital signal to produce fine data for summing with the coarse data and thereby form the output signal.

Trigonometric signal generator and machine control

Abstract: Digital and analog converter method and apparatus for generating trigonometrically related signals suitable for use with position measuring and position controlling systems. Two or more converter analog output signals are formed as a function of the digital input. The digital input generates a digital count difference between the counts in two digital counters. The two counters are both stepped by synchronously derived stepping pulses to produce counter output signals, exhibiting a phase difference proportional to the digital count difference. The counter output signals are logically combined to form analog output signals. Those analog output signals are pulse-width modulated rectangular waveforms which each include a fundamental sinusoidal frequency component having an amplitude proportional to a trigonometric function of the digital input. The analog output signals from the converter are typically connected as inputs to a position measuring device to trigonometrically define the position between two members of the position measuring device. The position measuring device is typically an transducer or other data element which responsively forms an analog output signal having a magnitude which indicates the relative position of the two members. The analog output signal is typically converted to a digital signal, in the form of a train of pulses, where each pulse represents an incremental distance. That digital signal, derived from the analog output signal, is typically supplied as an input to the converter which converts the digital input to the analog output, thereby forming a closed loop system.

System for reconstructing a digital signal

Abstract: A SYSTEM IS PROVIDED FOR REPRODUCING A TRANSMITTED SIGNAL HAVING ALTERNATE UPPER AND LOWER VOLTAGE LEVELS FROM A RECEIVED SIGNAL REPRESENTING THE TRANSMITTED SIGNAL CONTAMINATED WITH HIGH AND LOW FREQUENCY NOISE. A REFERENCE SIGNAL IS DEVELOPED HAVING A VOLTAGE LEVEL WHICH IS NOMINALLY MIDWAY BETWEEN THE UPPER AND LOWER VOLTAGE LEVELS OF THE TRANSMITTED SIGNAL, AND WHICH IS SHIFTED IN RESPONSE TO THE LOW FREQUENCY NOISE WITHIN THE RECEIVED SIGNAL. A LOCAL SIGNAL IS PRODUCED HAVING THE UPPER VOLTAGE LEVEL WHEN THE VOLTAGE LEVEL OF THE RECEIVED SIGNAL IS ABOVE THE VOLTAGE LEVEL OF THE REFERENCE SIGNAL, AND HAVING THE LOWER VOLTAGE LEVEL WHEN THE VOLTAGE LEVEL OF THE RECEIVED SIGNAL IS BELOW THE VOLTAGE LEVEL OF THE REFERENCE SIGNAL. THE LOCAL SIGNAL DUPLICATES THE TRANSMITTED SIGNAL.

System Design for Real-time Image Processing Based on Multi-core DSP

Abstract: In the process of real time image processing, the amount of image processing computing and data is very large, the use of single core or a single DSP can not meet the needs of real-time image processing. This paper designs the new generation multi-core DSP real-time image processing system, the system is a multi-core high performance DSP as the core circuit, and also uses FPGA counterweight device, high-speed data interface and high-speed external memory, to improve the data cache speed and data bandwidth, and the system satisfies the design of low loss and high computational performance. In order to verify the validity and reliability of the system, this paper uses the Hyper Lynx software to carry on the simulation test for the system, and sets the IP and sub net mask of image real time processing to obtain the eye and timing diagram of real-time image processing. Finally, the system image processing performance results can be obtained by the joint debugging, the simulation test results can be seen that the speed of real-time image processing is quick, and the rate of correct recognition is high, which can satisfy the need of real-time image processing, to provide the technical reference for the design of real time image processing system

Frequency domain analyzer using variable-rate time compression

Abstract: Contiguous frequency domain analysis of signals is achieved by sampling contiguous epochs of the signal; converting the sampled signal to a digital representation; storing the digital signal in a buffer; reading the stored digital signal a number of times at different reading rates to effect multiple frequency multiplications; converting the different digital signals red read analog equivalents; and passing the different analog signals through a fixed band-pass filter detector to effect an accelerated analysis of the frequency shifted signal.

Digital speech signal synthesizer

Abstract: Disclosed is an electrical-signal synthesizer for converting digitally coded information associated with at least one electrical signal, whose frequency, amplitude or phase may vary, to analog signals whose frequency, amplitude or phase varies in substantially the same manner as that of the at least one electrical signal. More specifically, the synthesizer is operative to convert a digital signal representative of a first analog signal, such as a voice signal, having varying parameters, such as frequency or amplitude, into an analog output signal which varies in substantially the same manner as the first signal, and where the digital signal is composed of consecutive frames of words, and one word of each frame is representative of a fundamental frequency associated with the first signal at an instant of time, and successive words in the respective frame are representative of the energy associated with at least one of a plurality of successive bands or spectrum segments of the first signal to be reproduced, at the given instant in time, each of the successive bands bearing a predetermined frequency relationship and wherein the synthesis of the output signal is accomplished by generating from the word representative of the fundamental frequency in each respective frame, a stream of digital words representative of the frequency and each of its harmonics at each instant of time and producing therefrom a second stream of digital words which is indicative of the frequency components of the original sound and modulating the second stream with amplitude data corresponding to discrete periods of time and adding the respective digital signals so produced for a discrete period of time and converting the same to an analog signal which is representative of the original voice signal.

Digital cathode-ray tube linearity corrector

Abstract: A system for correcting the signals transmitted to the deflection yokes of a cathode-ray tube employs digital means to produce an undistorted linear display while requiring a minimum of adjustments. Basically, the system utilizes digital circuitry to produce a correction factor from digital X- and Y-coordinate data supplied by a digital computer. This correction factor, which is equivalent to the sum of the squares of the X- and Y-coordinate data, is converted into an analog signal, along with the coordinate data, by digital to analog converters. The analog correction signal is multiplied by and then added to the analog coordinate signals, thereby producing the corrected deflection signals for an undistorted linear display on the essentially flat face of the tube. An analog correction signal may also be utilized to correct the focus of the electron beam on the tube face.

Hybrid implementation for sampled-data controllers

Abstract: A hybrid implementation for generalized digital filters in canonical form is presented, which uses analog elements for multiplication and summation and digital elements for storage (time delay). The hybrid controller is designed to realize virtually any compensation function up to three zeroes over three poles in the z domain with any sampling frequency up to several thousand hertz. Experimental results are presented for an example system.

The representation of continuous speech with a periodically sampled orthogonal basis

Abstract: The efficient transmission or processing of speech requires that a compromise be made between quality and bandwidth. Systems for bandwidth reduction, such as the vocoder, are usually designed to preserve the spectral content of the signal. High-quality systems, on the other hand, generally preserve waveshape by using high digital sampling rates. The determination of an adequate compromise is seriously impeded by the basic differences in these two approaches. The objective here is to investigate an analysis-synthesis procedure, that has been used to represent other signals, as a vehicle for determining this compromise. The continuous speech is divided arbitrarily into time periods and each period is expressed as a set of coefficients of an exponential expansion. The distinctive nature of speech is reflected in the choice of basis and analysis period rather than by special processing operations such as the pitch extraction of a vocoder. It has been demonstrated by digital simulation that with a proper selection of parameters both temporal waveshape and the spectrum can be preserved by this method. The statistically selected basis consists of ten pairs of damped sines and cosines and the experimentally chosen analysis period is 5.2 milliseconds. The coefficients of this expansion were measured by digital filtering on the computer. The simulated system is capable of synthesizing high-quality speech for speakers whose average pitch varied from 80 to 245 Hz without changing either the basis or the period. Although the feasibility of such a system has been demonstrated, a detailed investigation of coding techniques will be necessary before its efficiency can be compared to other approaches.

Image Processing With In-Line Optical Systems

Abstract: Image processing with in-line optical systems has been improved with improved experimental techniques and with precise methods of filter generation. Both analog and digital processing techniques are available and combined hybrid processing methods have been developed. Analog processing includes in-line and holographic optical processing and hybrid processing utilizes the combined capabilities of digital and analog systems. This paper is concerned with in-line optical processing and one hybrid scheme utilizing digital processing for filter fabrication and an in-line analog system for image processing. Experimental results are demonstrated with illustrations including restored defocused and blurred imagery.© (1969) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Data processing method for a weak, moving telemetry signal

Abstract: Method of processing data from a spacecraft, where the carrier has a low signal-to-noise ratio and wide unpredictable frequency shifts, consists of analogue recording of the noisy signal along with a high-frequency tone that is used as a clock to trigger a digitizer.

Laser-Acoustic Microwave Signal Processing

Abstract: The variable delay and processing of RF and microwave signals is a current problem of considerable importance both for the many military applications in RADAR and electronic countermeasures and for the civilian applications in signal processing. The very low velocity of acoustic waves, as compared to the propagation velocity of electromagnetic signals, permits the fabrication of physically compact systems capable of storing and processing electromagnetic signals. For example, a 1 /spl mu/sec delay in a non-acoustic delay line might require 700 feet of coaxial cable as compared to the 1/2 cm of material required in an acoustic delay line. While acoustic delay lines have fulfilled many of the needs of fixed delay lines, adequate continuously variable delay lines are not presently available. Among the several suggestions for obtaining a continuously variable delay, the use of the laser-acoustic interaction has been shown to be among the most promising. This paper will discuss the principles of such a laser-acoustic delay line and the extension of this device to a microwave signal processing device for performing either time compression, expansion and reversal or pulse compression, expansion and reversal.