Showing papers on "Digital signal processing published in 1986"

Cathedral-II: A Silicon Compiler for Digital Signal Processing

Abstract: The article describes the status of work at IMEC on the Cathedral-II silicon compiler. The compiler was developed to synthesize synchronous multiprocessor system chips for digital signal processing. It is a continuation of work on the Cathedral-I operational silicon compiler for bit-serial digital filters. Cathedral-II is based on a ?meet in the middle? design method that encourages a total separation between system design and reusable silicon design. The CAD system includes a rule-based synthesis program, a procedural program, and a controller synthesis environment. Processors are synthesized in terms of modules called from automated reusable module generators. Chip layout is done on a floor planner. An expert subsystem verifies correctness during silicon design and generates functional and timing models for verification at the module and chip levels.

Video transmission system

Abstract: A subscriber cable television system uses predominantly digital signal processing techniques and has extremely high security and an increased capacity for transmitting program and customer data to individual decoder units. For ease of data handling, two-channel audio, video, and high capacity program and customer data are multiplexed for transmission on the composite video signal. The decoder unit employs a system timing circuit which precisely synchronizes the sample times on the received composite video signal to the chroma burst, regardless of whether the video information is for a color or black-and-white program. An improved time-warp and segment scrambling method is disclosed along with means for suppressing the undesirable effects of discontinuities in the scrambled video signal. The digital audio is transmitted as scrambled most significant bits in low resolution samples and unscrambled least significant bits in a high resolution remainder sample. The system timing circuit has a horizontal sync detector accommodating variable line length such as is provided by some video recording apparatus. The clock to the horizontal counter is selectively phase-reversed in response to early or late horizontal sync so that the timing resolution is twice the clock period. An improved self-adjusting threshold detector and other means are disclosed for detecting a "20 IEEE" suppressed horizontal sync so that the full range of video modulation may be used more effectively. Circuitry is also disclosed for transmitting the customer and program information in a multi-level correlative signalling format in order to more effectively use the band width of the entire television channel.

Fast algorithms for digital signal processing

Abstract: This new textbook by R. E. Blahut, which deals with the theory and design of efficient algorithms for digital signal processing, contains perhaps the most comprehensive coverage of fast algorithms todate.Alargecollectionofalgorithmsistreated,withanemphasis on implementing the two canonical signal processing operations of convolution and discrete Fourier transformation. In recent years, there has been much work done on fast algorithms,andBlahutdoesafinejobofblendingmaterialfromdiverse sources to form a coherent and self-contained approach to his subject.The mathematical level of this book is high, reflecting the rather abstract nature of the theoretical underpinnings of fast computational techniques. Although electrical engineers are forthe most part mathematically sophisticated, they tend to lack training in abstract algebra and number theory, both of which are essential to any thorough discussion of fast algorithms. Thus this audience should find the tutorial chapters which the text provides on these topics to be quite helpful. An additional feature of the text, which the nonspecialist should find useful, is that each new algorithm is described through three different formats: a simple example, a flowchart, and a set of matrix equations. This use of repetition assists the reader in grasping subject matter which for the most part is nonintuitive. Operation counts (as measured by the number of multiplications and the number of additions) for each algorithm are tabulated for avarietyof blocklengths (i.e., lengths of data segments), making performance comparisons easy. As the author points out, run-time comparisons may be quite different. Each chapter concludes with a set of problems of varying difficulty. These problems are well integrated with the text and serve to supplement the many examples worked out in the text. The book is devoted to how one rapidly computes various mathematical operators such as transform and convolutions. For a deeper understanding of the meaning of theseoperators,one must consult other sources in which their use is discussed. The text emphasizes algorithms which employ a reduced, or minimum, numberof muItiplications,althoughadditioncountsarealsotaken into consideration. However, an algorithm which is the “fastest” as measured in arithmetic operation counts may not be the fastest in execution time, particularly if dedicated hardware is employed. Indeed in practiceother considerationsfrequentlypropel oneaway from the computationally ”optimal” algorithm. Much work has been done on the theory and application of signal processing algorithms which are “efficient” in terms other than rnultiply/add counts, such as roundoff noise, limit cycles, coefficient quantization, memory access, hardware costs, etc. It is clearly necessary to limit the scope of any treatise, and the exclusion of differing performance measures is certainly appropriate. A description of the contents of the book will now be given, followed by some concluding remarks of a more general nature. Chapter 2 i s a tutorial on abstract algebra. It i s quite readable and is liberally laced with examples. In addition to the standard modern algebra fare (groups, rings, fields, vector spaces, matrices), the ubiquitous Chinese remainder theorem is discussed in detail. Chapters 3 and 4, and their extensions in Chapters 7and 8, form the core of the text. The third chapter addresses fast algorithms for short convolutions. The Cook-Toom convolution algorithm is discussed, followed by the Winograd convolution algorithm. A proof of the optimality of the Winograd algorithm, with respect to multiplications, for performing cyclic convolutions, is presented at the close of the chapter. The fourth chapter addresses fast algorithms for computing the discrete Fourier transform. The CooleyTukey algorithm is considered first. The approach taken is to view this algorithm as a means of mapping a onedimensional Fourier transform into a multidimensional transform. Variations of the algorithm are discussed, including the Rader-Brenner transform. Next, the Good-Thomas algorithm is discussed. This algorithm is again presented as a means of mapping a onedimensional transform into a higher dimensional transform, this time based on the Chinese remainder theorem. Rader’s algorithm for computing primelength Fouriertransforms by useofconvolution ispresented next. Extensions of the algorithm to blocklengths which are the power of an odd prime are considered. The chapter closes with the Winograd-Fourier transform which builds upon the Rader prime algorithm. Certain short blocklengthsareconsidered in detail,and the corresponding algorithms are compiled into an Appendix. Chapter 5 i s a mathematical interlude, tutorially covering items from number theory and algebraic field theory which are needed in later chapters. Topics include the totient function, Euler’s theorem, Fermat’s theorem, minimal polynomials, and cyclotomic polynomials. Chapter 6 is devoted to number theoretic transforms. These transforms proceed by representing the data values themselves in the field of integers modulo a prime. Convolution in integer fields is also covered. Chapters 7and 8 extend the convolution and transform methods of Chapters 3 and 4 to higher dimensions. Multidimensional transforms (convolutions) are used both to efficiently compute onedimensional transforms (convolutions) and to process data which are inherently higher dimensional. Both applications are treated in these chapters. Topics include the Agarwal-Cooley convolution algorithm, polynomial transforms, the family of Johnson-Burrus transforms and the Nussbaumer-Quandalle FFT. Chapter 9 discusses architectures for transforms and digital filters and includes treatmentsof FFT butterfly networks and overlapadd convolution. The remaining three chapters are mostly independent from the rest of the book. Chapter 10 covers fast algorithms based on doubling strategies. Computational tasks for which such fast algorithms are derived include sorting, matrix transposition, matrix multiplication, polynomial division, computation of trigonometric functions, and coordinate rotation. Many of theseoperations arise as steps in the solution of oneor more signal processing problems. Fast algorithms for solving Toeplitz systems is the theme of Chapter 11. There is a variety of fast algorithms discussed, the proper choice of which depends on the specific structure of the Toeplitz system at hand (such as whether or not the system is symmetric and whether or not the right-hand vector is arbitrary). The final chapter addresses fast algorithms for Trellis and tree search and includes the Viterbi, Stack, and Fano algorithms. These

Signal Processing - The Model-based Approach

Abstract: Signal processing: the model-based approach , Signal processing: the model-based approach , مرکز فناوری اطلاعات و اطلاع رسانی کشاورزی

Audio and video digital recording and playback system

Abstract: A microcomputer system for converting an analog signal, such as an audio or video signal representative of sound or video into a digital form for storing in digital form in a highly condensed code and for reconstructing the analog signal from the coded digital form. The system includes reductive analytic means where the original digital data stream is converted to a sequential series of spectrograms, signal amplitude histrograms and waveform code tables. Approximately 100 times less storage space than previously required for the storage of digitized signals is thereby obtained. Additive synthesis logic interprets the stored codes and recreates an output digital data stream for digital to analog conversion that is nearly identical to the original analog signal.

Adaptive filters suitable for real-time spectral analysis

Abstract: Some new cascade structures for adaptive filters are presented. They are especially suitable for real-time applications. Since the new structures are intended to be realized using single chip DSP IC's or single chip custom VLSI circuits the requirements for memory and divisions are minimized. The new structures are based on state-variable biquads that in addition to having good SNR's and low sensitivities (for fixed-point implementations) can also have their resonant frequencies and Q -factors independently tuned. The special cass of using the adaptive filters for tracking sinusoids corrupted by noise and for formant based speech compression are described in detail.

Geophysical signal processing

Abstract: Draws together a number of areas of knowledge to give unified coverage of the subject: the geophysical applications of digital signal processing. The presentation has a strong applications orientation. The coverage connects and unifies several fields, namely wave propagation, digital signal processing, spectral analysis, and computer methods. The book covers many topics in depth.

Apparatus for generating an image from a digital video signal

Abstract: An apparatus for generating a high quality image from a digital video signal includes a system for gamma correcting the digital video signal with a digital look up table and for converting the resultant digital signal to an analog video signal. Another circuit generates a triangular wave reference pattern signal and a comparator compares the analog video signal with the triangular wave reference pattern signal to form a pulse-width-modulated signal. A raster scanning print engine producing, for example, a laser beam, scans over a recording medium in accordance with the pulse-width-modulated signal, thereby forming an image of high quality on the recording medium of a print engine. This apparatus can also be used with an analog video signal by first converting the analog video signal to a digital video signal with an analog to digital converter.

Adaptive nonlinear digital filters using distributed arithmetic

Abstract: A nonlinear adaptive filter structure, based upon the theory of the truncated discrete Volterra series, is presented. A memory-oriented implementation exploiting distributed arithmetic is considered, and the conventional LMS adaptation algorithms are suitably modified. Memory-size reduction methods are developed to obtain simpler actual realizations. Computer simulation results are presented.

Digital method and system for reproducing analog data

Abstract: There is disclosed herein a method of and system for reproducing an analog signal using digital techniques. The system includes means for encoding the analog signal in digital form and means for decoding the encoded analog signal. The encoding means includes means for detecting the upper bandwidth limit of the input analog signal during each of a plurality of successive sample intervals. Means coupled to the detecting means are included for sampling the input analog signal at a sampling rate determined in accordance with the detected upper bandwidth limits to derive a series of voltage levels which together define a sampled approximation of the input analog signal. Means are coupled to the sampling means for converting each voltage level and the sampling rate at which such voltage level was derived into digital signals. Also included are means for detecting when the input analog signal is at substantially a zero level, means for developing a digital indication of the length of time the signal is at such level and means for combining the digital indication with the digital signals to develop a digital representation of the input analog signal. The decoding means includes means for sequentially retrieving the digital signals, means coupled to the retrieving means for converting the digital signals into the sampled approximation of the input analog signal and a low-pass filter coupled to the converting means having a cut-off frequency determined in accordance with the encoded sampling rate for filtering the sampled approximation to reproduce the input analog signal.

Digital transmisson including add/drop module

Abstract: In a transmission system, adding and/or dropping any one or more of a plurality of digital signals of one or more digital transmission bit rates is facilitated by employing a un- lque transmission signal in which data words associated with Individual digital signals are arranged in prescribed groups. The transmission signal data word groups are obtained by formatting the individual digital signals to be combined into a unique channel frame format common to all of the digital signals and by employing a unique one-step multiplexing process to insert digital words from the channel frames into the group of data word positions in the transmission signal associated with the particular signal being combined. Consequently, digital signals may be added to the transmission signal by formatting them into the common channel frame format and, then, inserting the digital words therefrom in the one-step multiplexing process into an associated group of data words positions in the transmission signal. Digital signals are dropped from the transmission signal by extracting associated groups of data words from the transmission signal, identifying the corresponding channel frames and deformatting the data bits from channel frames of corresponding digital signals being reconstructed.

The Motorola DSP56000 Digital Signal Processor

Abstract: The DSP56000 brings 10.25-MIPS performance to digital signal processing and retains enough similarities to other Motorola microprocessors to make it easy to learn and program.

Fault-Tolerant Systems For The Computation Of Eigenvalues And Singular Values

Abstract: The computations of eigenvalues and singular values are key to applications including signal and image processing. Since large amounts of computation are needed for these algorithms, and since many digital signal processing applications have real-time requirements, many different special-purpose processor array structures have been proposed to solve these two algorithms. This paper develops a new methodology to incorporate fault tolerance capability into processor arrays which have been proposed for these problems. In the first part of this paper, earlier techniques of algorithm-based fault tolerance are applied to QR factorization and QR iteration. This technique encodes input data at a high level by using the specific property of each algorithm and checks the output data before they leave the systems. In the second part of the paper, special properities of eigenvalues and singular values are used to achieve the error detection without encoding the input data. Fault location and reconfiguration are performed only after an erroneous signal has been detected. The introduced overhead is extremely low in terms of both hardware and time redundancy.

Modifications to the McClellan, Parks, and Rabiner computer program for designing higher order differentiating FIR filters

Abstract: Simple modifications to the McClellan, Parks, and Rabiner linear phase finite impulse response (FIR) filter design program are suggested to allow the design of an nth-order differentiating FIR filter of arbitrary length for any n. Two illustrative examples are also provided.

A New Approach to E.H.V. Direction Comparison Protection Using Digital Signal Processing Techniques

Abstract: Some problems concerning the performance of superimposed component directional relays are discussed, and it is shown that the basic directional criteria can be violated due to travelling wave phenomena or for fault occurrence approaching a voltage zero. A relay design is developed using digital signal processing techniques to overcome these problems. Tlhe complete relay is tested using a typical 40OkV transmission line application.

The Devevolpment of a New High Speed 3-Terminal Line Protection Scheme

Abstract: A new high speed Current Differential protection scheme for application to three-terminal transmission lines is described. The scheme utilises a wide-band Fibre Optic Link and the results show that the special filtering and signal processing techniques developed provide a performance which satisfies the requirements for reliable and secure protection of Teed feeders. The protection was developed using CAD techniques and the methods proposed are readily implemented using present generation signal processing hardware.

Optical crossbar interconnected digital signal processor with basic algorithms

Abstract: A proposed high performance gigaflop signal processor is described in which 512 processors are interconnected with a 768 by 768 crossbar switch utilizing a spatial light modulator of the type presently under development at Texas Instruments. Optical fibers are used to provide high speed communication between the processors and the switch. The system, processor nodes, programming, and functional operation are described. The advantages are discussed for reconfigurability, optical crossbar switches, and programmed data flow. Efficient implementations are presented for a systolic filter, a fast Fourier transform, a correlator, and a matrix-vector multiplier. Preliminary performance estimates suggest that over one gigaflop performance is achievable for these algorithms on this processor.

Modulated digital radio frequency memory

Abstract: A digital radio frequency memory (DRFM) system for storing and retrieving radio frequency signals in a digital memory. The system includes a digital memory, digital converters at the input and output of the digital memory, and input and output mixers. The RF signal to be stored is applied to the input mixer along with a modulated local oscillator signal to produce an IF signal which is converted into digital values for storage in the digital memory. When the stored digital values are retrieved from the memory to reproduce the RF signal, they are converted back into an analog IF signal which passes through a low-pass filter before being applied to the output mixer. The local oscillator signal, modulated with the same waveform used in storing the digital values, is also applied to the output mixer to provide the reconstructed RF signal. A pseudo random, discrete digital valued waveform is used to phase modulate the local oscillator signal in the preferred embodiment of the invention. This arrangement permits spur free, image free reproduction of RF signals over a large bandwidth.

The role of digital signal processing in NDT

Abstract: Non-destructive testing is playing an increasingly important role in a number of industries. Signal processing techniques have significantly advanced the state-of-the-art of non-destructive testing and evaluation. This paper briefly reviews the role of signal processing in NDT. It has dual purpose: firstly, to review a broad range of applications of signal processing to NDT and secondly, to present a brief overview of signal processing approaches used for detect characterization. This deals with signal acquisition and problems associated with sampling and quantization, signal enhancement and information retrieval, with emphasis on application of pattern recognition techniques to defect detection and characterization.

Design of high-speed digital filters suitable for multi-DSP implementation

Abstract: A multipath signal processing scheme is proposed to overcome the limitation on the throughput rate of present day available LSI devices which is specifically suitable for implementation using a number of Digital Signal Processors (DSP). Two methods are proposed to realize a given transfer function H(z) of digital filter, with a throughput rate speed up factor of N , over the conventional methods. The first method, called Delayed Multipath Approach here, uses an N -path structure as a building element. These N elements are connected successively with increasing delay units to realize a given transfer function. The second method preprocesses the input signal sequence by an FFT processor and follows it up by N of constituent transfer functions derived from H(z) having real coefficients. The output of these N constituent transfer functions are finally postprocessed by inverse FFT processor to obtain the desired output signal. The number of the constituent transfer functions are double for a special case when the transfer function to be implemented has complex valued coefficients. These two methods serve as complementary approaches, because the first method is better suited for small values of the speed-up factor N and the second one has distinct advantage for larger values of N . The discussion of the first design method is organized in two parts: FIR filter design and IIR filter design, for each of which 2-path and N -path structures are separately explained. The second design method is discussed under the headings of real transfer function and complex transfer function. Design examples are also given to illustrate both of these two methods. Finally, a multi-DSP hardware system is outlined which is specifically designed for implementing the multipath structures discussed here.

Adaptive filters suitable for real-time spectral analysis

Abstract: New cascade structures for adaptive filters are presented. They are especially suitable for real-time applications. Since the new structures are intended to be realized using single-chip DSP ICs or single-chip custom VLSI circuits, the requirements for memory and divisions are minimized. The new structures are based on state-variable biquads that in addition to having good SNRs and low sensitivities (for fixed-point implementations) can also have their resonant frequencies and Q-factors independently tuned. The special cases of using the adaptive filters for tracking sinusoids corrupted by noise and for formant based speech compression are described in detail.