Showing papers on "Digital signal processing published in 1978"

Digital Processing of Speech Signals

Abstract: 1. Introduction. 2. Fundamentals of Digital Speech Processing. 3. Digital Models for the Speech Signal. 4. Time-Domain Models for Speech Processing. 5. Digital Representation of the Speech Waveform. 6. Short-Time Fourier Analysis. 7. Homomorphic Speech Processing. 8. Linear Predictive Coding of Speech. 9. Digital Speech Processing for Man-Machine Communication by Voice.

Cubic splines for image interpolation and digital filtering

Abstract: This paper presents the use of B-splines as a tool in various digital signal processing applications. The theory of B-splines is briefly reviewed, followed by discussions on B-spline interpolation and B-spline filtering. Computer implementation using both an efficient software viewpoint and a hardware method are discussed. Finally, experimental results are presented for illustrative purposes in two-dimensional image format. Applications to image and signal processing include interpolation, smoothing, filtering, enlargement, and reduction.

Applications of digital signal processing

Abstract: Applications of digital signal processing , Applications of digital signal processing , مرکز فناوری اطلاعات و اطلاع رسانی کشاورزی

Digital Processing Techniques in the 60 Channel Transmultiplexer

Abstract: Techniques applied to the digital processing of speech and signaling in a 60 channel FDM to 30 channel PCM converter are presented. Great efficiency is achieved through connecting a particular type of Fourier Transform computer to a polyphase network in which coefficient symmetries are exploited. Digital processing parameter values have been chosen so as to meet specified performance objectives. Realization of a laboratory model is outlined and measured performance is reported. Adequacy of the various options retained for the digital processing is confirmed by the experimental results.

Real-time digital x-ray subtraction imaging

Abstract: Difference images, derived from an X-ray image of an anatomical subject, are produced in real time by directing X-rays through the anatomical subject to produce an X-ray image, converting the X-ray image into television fields comprising trains of analog video signals, converting the analog video signals into digital video signals, producing integrated mask digital video signals by integrating the digital video signals over a mask time interval, subtracting the integrated mask digital video signals from corresponding digital video signals of television fields subsequent to the mask time interval and thereby producing digital difference video signals, converting the digital difference video signals into analog difference video signals, and converting the analog difference video signals into a series of visible television difference images representing changes in the X-ray image subsequent to the mask time interval. The mask time interval preferably corresponds generally to at least one complete cardiac cycle of the anatomical subject. An X-ray contrast medium is preferably injected into a peripheral blood vessel of the anatomical subject, with a timing such that the contrast medium appears in the X-ray image subsequent to the mask time interval. In another embodiment, the integrated mask digital video signals are reconverted to analog form and are subtracted on an analog basis from the analog video signals produced subsequent to the mask time interval, to produce the analog difference video signals.

Operations on delta-modulated signals and their application in the realization of digital filters

Abstract: A new method is proposed with which it is possible to find a delta-modulated signal of the half sum of two analogue signals through a direct operation on their delta-modulated form. With a direct operation it is also possible to form a delta-modulated signal of the product of an analogue signal by a constant. The resulting modulated signal includes an error which is generally considered negligible. The hardware implementation of the method is both, simple and modular employing only conventional full adders and D-flip-flops. This leads to the possibility of realization of digital filters with straight delta-modulated input and output signals which are not intermediately transformed. Multiplexing for changing coefficients is also possible.

High-Speed Monolithic Multipliers for Real-Time Digital Signal Processing

Abstract: Real-time digital signal processing requires very fast multiplication, which is now becoming possible using mathematical techniques to take advantage of single-chip multipliers.

Relationships between digital signal processing and control and estimation theory

Abstract: The purpose of this paper is to explore several current research directions in the fields of digital signal processing and modern control and estimation theory. We examine topics such as stability theory, linear prediction, and parameter identification, system synthesis and implementation, two-dimensional filtering, decentralized control and estimation, and image processing, in order to uncover some of the basic similarities and differences in the goals, techniques, and philosophy of the two disciplines.

Real-time digital X-ray time interval difference imaging

Abstract: Difference images, derived from an X-ray image of an anatomical subject, are produced in real time by directing X-rays through an anatomical subject to produce an X-ray image, converting the X-ray image into a series of television fields comprising trains of analog video signals, converting the analog video signals into corresponding digital video signals, integrating the digital video signals over a series of successive time intervals corresponding with a plurality of television fields and thereby producing a series of sets of integrated digital video signals, performing a series of subtractions between each set of integrated video signals and the preceding set of integrated video signals and thereby producing a series of successive digital difference video signals, converting the digital difference video signals into analog difference video signals, and converting the analog difference video signals into a series of visible television difference images representing changes in the X-ray image between the successive time intervals. Apparatus for carrying out this method is also disclosed.

Cubic Splines for image Interpolation and Filtering

Abstract: This paper presents the use of B-splines as a tool in various digital signal processing applications. The theory of B-splines is briefly reviewed, followed by discussions on B-spline interpolation and B-spline filtering. Computer implementation using both an efficient software viewpoint and a hardware method are discussed. Finally, experimental results are presented for illustrative purposes in two-dimensional image format. Applications to image and signal processing include interpola- tion, smoothing, filtering, enlargement, and reduction.

System and method for speech recognition

Abstract: A system and method for speech recognition provides a means of printing phonemes in response to received speech signals utilizing inexpensive components. The speech signals are inputted into an amplifier which provides negative feedback to normalize the amplitude of the speech signals. The normalized speech signals are delta modulated at a first sampling rate to produce a corresponding first sequence of digital pulses. The negative feedback signal of the amplifier is delta modulated at a second sampling rate to produce a second sequence of digital pulses corresponding to amplitude information of the speech signals. The speech signals are filtered and utilized to produce a digital pulse corresponding to high frequency components of the speech signals having magnitudes in excess of a threshold voltage. A microprocessor contains an algorithm for detecting major slope transitions of the analog speech signals in response to the first sequence of digital signals by detecting information corresponding to presence and absence of predetermined numbers of successive slope reversals in the delta modulator producing the first sequence of digital pulses. The algorithm computes cues from the high frequency digital pulse and the second sequence of pulses. The algorithm computes a plurality of speech waveform characteristic ratios of time intervals between various slope transitions and compares the speech waveform characteristic ratios with a plurality of stored phoneme ratios representing a set of phonemes to detect matching therebetween. The order of comparing is determined on the basis of the cues and a configuration of a phoneme decision tree contained in the algorithm. When a matching occurs, a signal corresponding to the matched phoneme is produced and utilized to cause the phoneme to be printed. In one embodiment of the invention, the speech signals are produced by the earphone of a standard telephone headset.

Digital apparatus approximating multiplication of analog signal by sine wave signal and method

Abstract: A method and an apparatus are disclosed which employ digital techniques to approximate a multiplication of an analog signal by a sine wave of an appropriate frequency for detecting the analog signal The analog signal is converted to a digital representation which includes a binary sign bit and a plurality of binary magnitude bits The sign bit of the digital representation is selectively complemented by an inverting logic gate which operates under the control of a square wave having the appropriate frequency of the sine wave which is being approximated The selective complementation of the sign bit of the digital representation functions to multiply the digital representation by +1 or -1 as determined by the status of the square wave signal The resulting digital representation is latched at periodic intervals for further processing of the detected signal which results from the quasi digital multiplication

On decoding techniques for residue number system realizations of digital signal processing hardware

Abstract: In a recent issue of this journal, a hardware implementation of the Chinese Remainder Theorem was proposed for the translation of residue coded outputs into natural integer for an FIR filter realization. The method requires a modulo M adder-shifter network which is not commercially available and has to be constructed from more basic logic elements. This letter presents a more successful technique based on a mixed-radix conversion process. The residue to binary decoder can be built using commercially available elements and requires shorter word lengths in the ROM which stores the output function.

Time Integrating Optical Processors

Abstract: Over the past few years there has been an exponential increase in interest in analog processing technology. This is largely due to economics. The cost of digital processing at very high data rates is often prohibitive. In addition, many of the components required for optical processing have matured to the "off-the-shelf" stage. A significant portion of the effort in optical processing has shifted from components development to processing architecture. The time integration architecture offers the following advantages: a. It makes the most effective use of CCD arrays as sensors. b. It often produces a significant data rate reduction. (output rate is often orders cf magnitude below the input rate) c. It does not require a two dimensional electrooptic modulator to implement two dimensional operations. The following paper reviews past efforts in one dimensional time integrating processors and introduces the two dimensional time integrating correlator. This correlator can implement a variety of operations including ambiguity functions and large time bandwidth spectrum analysis (two dimensional).

Digital delay/ambience processor

Abstract: A processor for audio program material which synthesizes simulated reflection and reverberation signals for use in creating the illusion of a specific modeled listening environment. A preferred embodiment of the processor accepts stereophonic audio signals and simulates the multiple reflections of an acoustically modeled environment by recirculating the signals through a pair of digital signal delays which have different delay times. The analog input signals are sampled and converted to digital form by first attenuating the signals with a digitally controlled step attenuator and then encoding the amplitude of the attenuated signal using a ramp comparison technique. The encoded amplitude signals, and a code representative of the attenuator position, are stored in a random access memory and are retrieved after the desired delay. The retrieved digital data are converted back to analog signals by reversing the encoding process, using the same ramp voltage generator. A stack comprised of a counter and two registers connected in a loop generates three addresses during each sample period for use in writing in new data and retrieving previously stored data.

Real time digital sound synthesizer

Abstract: Linearly binary-coded digital control signals representing specific time segments of musical sounds are used in part to control stepped ramp signal generators in a digital synthesizer which runs continuously on a fixed program. The ramp signals are used to control amplitude and frequency parameters of multiple digital oscillators that produce respective constituent tones of the musical sound segments. The synthesizer is operable in response to time-multiplexed digital control signals for multiple musical voices, one voice portion of which is also being computed in real time for multiplexing with previously computed and stored digital control signals for other voices. The indicated computations are effected by known techniques on a commercially available computer to translate performer-actuated transducer output signals, designating notes of a composition, into the aforementioned binary-coded digital control signals.

An approach to the implementation of digital filters using microprocessors

Abstract: The design of digital filters using microprocessors is discussed along with simple solutions to improve the speed and memory requirements of such systems.

Circuit arrangement for correcting digital signals

Abstract: The analog output signals of a photoelectric image converter or transducer are passed through an analog-to-digital converter and then corrected to take various irregularities of the optical sensor elements of the transducer into account. A reference signal and a dark signal are subtracted from the individual image point signals. The difference is then multiplied by a factor representing the slope of the characteristic response curve of the respective optical sensor element. The so corrected signal may be used for further processing or it may be subjected to a further correction operation.

Wheel speed sensing system

Abstract: A wheel speed sensing system for a brake control system is characterized in that the periodic signal generated by a wheel speed transducer is used to trigger a series of digital signals having a frequency proportional to that of the periodic transducer signal. Clock signals are counted between a selected pair of digital signals to provide a digital measure of the time interval separating the digital signals and hence the period of the periodic signal. In a preferred embodiment the number of intervening digital signals separating the selected pair of digital signals is automatically chosen to maintain the time resolution of the measurement above a preselected value.

Digital TDM-FDM Translator with Multistage Structure

Abstract: In this paper, a new digital signal processing algorithm for the digital TDM-FDM translator is proposed. The digital TDM-FDM translator, which performs a direct translation between two multiplex formats in the telephone network (time-division-multiplexing (TDM) and frequency-division-multiplexing (FDM)) by using digital techniques, has advantages in accuracy and stability of characteristics over equivalent analog equipments. However from the economical point of view, it largely depends on the cost reduction of semiconductor devices and LSI technologies. The proposed algorthm can be realized using only two digital filters and does not require product modulators or Fast Fourier Transform (FFT) processors. The required number of multiplications, which is closely related to the quantity of hardware, is considerably reduced by the multistage structure of this algorithm. The reduction in the kind of required digital hardware and the required number of multiplications makes it possible to efficiently utilize the new hardware realization techniques of digital filters or multipliers using read-only memories and simple logic devices. Since it is foreseen that cost reduction of memory devices will be more rapid than that of logic devices, the proposed algorithm is expected to be advantageous with regard to cost over existing algorithms where complex multiplier logic is required. The estimation of the computation rate is carried out with reference to a practical case. The computer simulation results are also shown.

Optical data processing : applications

Abstract: What do you do to start reading optical data processing applications ? Searching the book that you love to read first or find an interesting book that will make you want to read? Everybody has difference with their reason of reading a book. Actuary, reading habit must be from earlier. Many people may be love to read, but not a book. It's not fault. Someone will be bored to open the thick book with small words to read. In more, this is the real condition. So do happen probably with this optical data processing applications .

System for recording and/or reproducing an audio signal which has been converted into a digital signal

Abstract: A composite video signal recording and reproducing system also records and reproduces an audio signal after their conversion into a digital signal. The recording system converts an input analog audio signal into a digital signal and generates a synchronizing signal corresponding to a synchronizing signal of the composite video signal. There is a time-axis compression of the converted digital signal so that they do not exist in the period corresponding to the synchronizing signal. A level-setter sets the level of the digital signal which has been compressed at a value which is lower than the white peak level of the composite video signal. A mixer mixes the level set digital signal and the generated synchronizing signal, to obtain a composite digital signal. The composite digital signal is supplied to the recording and/or reproducing apparatus and recorded on a recording medium. The level-setter sets the amplitude of the digital signal at a magnitude such that the overshoots and undershoots occurring as a result of the passage of the digital signal through a pre-emphasis circuit of the recording and/or reproducing apparatus will not be clipped by a clipping circuit of the recording and/or reproducing apparatus.

A generalized scheme for an all-digital time-division multiplex to frequency-division multiplex translator

Abstract: A generalized scheme for an all-digital time-division multiplex (TDM) to frequency-division multiplex (FDM) translator is discussed. It is shown that existing systems for this translation, e.g., those in which a fast Fourier transform (FFT) processor is used, can be derived from this generalized scheme as special cases. Several new Implementations can also be derived from this scheme when different processors are used. A specific example using a Hadamard processor is discussed.