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A Method of Achieving Fully Automatic Frequency Management for HF Systems

A Method of Achieving Fully Automatic Frequency Management f...

引用

MILCOM, Military Communications conference

作者： Dan O. Weddle L. D. Witkemper Collins Defence Communications ROCKWELL INTERNATIONAL USA

This paper describes the automatic frequency management system (AFMS) approach which is currently being developed and tested by Rockwell to satisfy new and evolving requirements to make HF communications systems more self-adaptive to the channel requirements and less operator-intensive. Most modern HF systems currently utilize digital data modulation in conjunction with digitized or analog voice modulation to transfer information across secure or non-secured links. The established trend is toward all digital networks in which record traffic, facsimile, and synthesized or real time (digital) voice modes are accommodated by digital signal processing (DSP) techniques to transfer data in a significantly more robust fashion than was possible prior to the advent of DSP technology. The AFMS system described here will ultimately exploit this trend toward digitized data by extracting from normal communications the link quality information needed to measure the channel. The measurement data currently being tested is extracted from advanced link quality analyzer (ALQA) test transmissions and is reduced to a single MOF value for each measured link by a MOF determination algorithm. This algorithm allows n measured links to provide the MOF data necessary to adequately forecast the MUF on m unmeasured links, where n is much less than m. The forecasting technique is based on providing near-real time updates (pseudo SSN) to the well known MINIMUF 3.5 model. The measurement and forecasting process is carried out continuously and automatically allowing the updates to the in-place frequency plan.

关键词： Automatic frequency control Hafnium Digital signal processing Digital modulation Data mining Current measurement signal processing algorithms Automatic testing System testing Communication system traffic

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IMPACT OF advanced COMPUTATIONAL TECHNIQUES ON DIGITAL AVIONICS signal PROCESSOR PERFORMANCE.

IMPACT OF ADVANCED COMPUTATIONAL TECHNIQUES ON DIGITAL AVION...

引用

Proceedings of the AIAA/IEEE 6th Digital Avionics Systems conference.

作者： Vojir, William M. Grumman Aerospace Corp Bethpage NY USA Grumman Aerospace Corp Bethpage NY USA

The application of the Winograd technique and prime factor algorithm to the discrete Fourier transform is examined. The case of 24-point transform is used to illustrate the effect of overhead computational architectures. Attention is then given to more exotic algorithms and their possible impact on digital avionics design. The possibility of entirely eliminating multiplications from certain filtering operatings is then considered. Finally the impact of advanced CAE techniques on signal processing system design is analyzed.

关键词： signal processing

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Adaptive image smoothing algorithms for edge and texture preservation

Adaptive image smoothing algorithms for edge and texture pre...

引用

International conference on Acoustics, Speech, and signal processing (ICASSP)

作者： K. Tao ESL Inc. Sunnyvale CA USA

A class of reasonably simple and efficient adaptive algorithms are developed to enhance noise degraded images while preserving the edge and texture information. Techniques from one-dimensional adaptive signal processing and systems identification are extended and applied to two-dimensional image smoothing through proper modelling of the image. Both the AR and ARMA models are treated. The conceptual separation of "image causality" and "processing causality" is advocated. advanced topics that are covered include: faster computation algorithms, simultaneous contrast stretching and smoothing, etc.

关键词： Smoothing methods Predictive models signal processing algorithms Pixel Statistics Adaptive filters Java Adaptive algorithm Degradation Adaptive signal processing

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A performance comparison of pitch extraction algorithms for noisy speech

A performance comparison of pitch extraction algorithms for ...

引用

International conference on Acoustics, Speech, and signal processing (ICASSP)

作者： K. Oh C. Un Department of Electrical Engineering Korea Advanced Institute of Science and Technology Seoul South Korea

Results of a performance comparison study of eight pitch extraction algorithms for noisy as well as clean speech are presented. These algorithms are the autocorrelation method with center clipping, the autocorrelation method with modified center clipping, the simplified inverse filter tracking (SIFT) method, the average magnitude difference function (AMDF) method, the pitch detection method based on LPC inverse filtering and AMDF, the data reduction method, the parallel processing method and the cepstrum method. It has been found that for pitch detection of noisy speech the algorithm that uses an AMDF or an autocorrelation function yields relatively good performance than others. A pitch detector that uses center clipped speech as an input signal is effective in pitch extraction of noisy speech. In general, preprocessing such as LPC inverse filtering or center clipping of input speech yields remarkable improvement in pitch detection.

关键词： Detectors Autocorrelation Speech processing Speech analysis Detection algorithms Speech enhancement Acoustic noise Linear predictive coding Filtering Parallel processing

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A manpack portable LPC 10 vocoder

A manpack portable LPC 10 vocoder

引用

International conference on Acoustics, Speech, and signal processing (ICASSP)

作者： B. Fette C. Rimpo J. Kish Motorola Government Electronics Group Scottsdale AZ USA

A manpack portable LPC-10 Vocoder has been developed which makes substantial size and power performance improvements over existing LPC Vocoders. This is accomplished through the use of distributed processing using the latest VLSI digital signal processing technology and advanced microcomputer technology. First the LPC-10 standard algorithm was partitioned into high performance digital signal processing tasks that may stand alone as VLSI devices. The remaining LPC-10 algorithmic components are partitioned by the data and process flow graphs into meaningful multi-purpose stand alone single chip computers, resulting in a vocoder that uses 3 VLSI, and 3 LSI components. The digital signal processing algorithms are partitioned as follows: LPC Analysis IC, LPC Synthesis IC, AMDF Pitch Extraction IC. The data flow processes are partitioned into microcomputers as follows: Transmit Pitch and Voicing in processor #1, Transmit AGC in processor #2, and Parameter Quantization and Serialization in processor #3; in the receive mode sync acquisition and maintenance and parameter deserialization in processor #3, error correction and dequantization in processor #2, and interpolation rule implementation in processor #1. The data flow processor's partitions were greatly affected by the use of single chip computers. These computers have a very limited RAM and ROM space causing the partition to be dependent on program size. The use of single chip computers minimizes external hardware necessary for the vocoder implementation.

关键词： Linear predictive coding Vocoders Very large scale integration Digital signal processing chips Partitioning algorithms signal processing algorithms Integrated circuit synthesis Digital integrated circuits Microcomputers Data flow computing

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Error analysis of LMS adaptive digital filter implemented with logarithmic number system

Error analysis of LMS adaptive digital filter implemented wi...

引用

International conference on Acoustics, Speech, and signal processing (ICASSP)

作者： V. Shenoy F. Taylor Electrical and Computer Engineering Department University of Cincinnati Cincinnati OH USA Advanced Micro Devices Inc. Sunnyvale CA USA Electrical Engineering Department University of Florida Gainesville FL USA

This paper considers implementation of a least mean square (LMS) adaptive digital filter algorithm using logarithmic number system (LNS). Analytical expressions for error performance have been derived and design issue... 详细信息

关键词： Error analysis Least squares approximation Digital filters Adaptive filters Quantization Algorithm design and analysis Equations Design engineering signal processing algorithms Performance analysis

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A 16-bit three port arithmetic logic and shift unit

A 16-bit three port arithmetic logic and shift unit

引用

International conference on Acoustics, Speech, and signal processing (ICASSP)

作者： J. Nuttall J. Oxxal Analog devices Inc. Norwood Massachusetts

The DSP division of Analog Devices is producing a CMOS, 75ns, 16 bit Arithmetic, Logic, and Shift Unit (ALSU). The ADSP-1200 features a 16-bit wide ALU, three ports, on board barrel shifter in parallel with the ALU, versatile register files, priority encoder and highly fielded instruction set. The device was specifically optimized to efficiently handle algorithms typically found in DSP. The first section of this paper gives an overview of the ADSP-1200. The second section describes the architecture and functionality of the device. Finally, the instruction set is briefly described. The ADSP-1200 shows that high speed CMOS combined with advanced packaging techniques results in effective, producible devices for DSP problems.

关键词： Arithmetic Digital signal processing CMOS logic circuits Logic devices Testing Automatic control Logic arrays Packaging CMOS process signal processing

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ON CONVERGENCE PROPERTIES OF ADAPTIVE algorithms FOR signal processing.

ON CONVERGENCE PROPERTIES OF ADAPTIVE ALGORITHMS FOR SIGNAL ...

引用

1983 Proceedings of the International conference on Systems, Man and Cybernetics.

作者： Murali, T. Rao, B.V. Indian Inst of Technology Advanced Cent for Research in Electronics Bombay India Indian Inst of Technology Advanced Cent for Research in Electronics Bombay India

Adaptive signal processing algorithms have been classified and their convergence behavior analyzed. Stochastic approximation is used as the basis for the classification into first-order stochastic approximation, second-order stochastic approximation, and nonstochastic approximation algorithms. First-order stochastic approximation algorithms are easy to implement compared to second-order algorithms, which are computationally expensive. However, from the convergence point of view, second-order stochastic approximation algorithms are superior. Two methods of convergence analyses, the eigenvalue method and the ordinary differential equation method, are discussed. A few results in the form of learning curves obtained from computer simulation studies are presented.

关键词： signal processing

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advanced ESM SYSTEM WITH AUTOMATIC processing FOR ELECTRONIC DEFENSE APPLICATIONS.

ADVANCED ESM SYSTEM WITH AUTOMATIC PROCESSING FOR ELECTRONIC...

引用

conference Proceedings - Electronics for National Security.

作者： Pope, Gregory M. EM Systems Inc Sunnyvale CA USA EM Systems Inc Sunnyvale CA USA

In the high density signal environments postulated for future military hostilities, the acquisition and identification problem facing today's Electronic Warfare (EW) planners is formidable. EMS system designs of the future will dictate receivers of wide instantaneous RF bandwidth coupled to automatic signal processors exhibiting powerful processing algorithms in order to overcome high received pulse rates and provide a real-time response. The design and performance characteristics of an advanced ESM system which incorporates wide instantaneous bandwidth receiving and real-time automatic processing/decision making techniques to provide rapid acquisition, classification, and identification of emitters in dense signal environments is described.

关键词： signal DETECTION

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Generalized CORDIC for digital signal processing

Generalized CORDIC for digital signal processing

引用

1982 IEEE International conference on Acoustics, Speech, and signal processing, ICASSP 1982

作者： Lee, Daniel T.L. Morf, Murtin Hewlett-Packard Laboratories 1501 Page Mill Road Palo AltoCA94304 United States Information Systems Laboratory Stanford University StanfordCA94305 United States

Many fast signal processing algorithms are based on a geometric framework and efficient implementations of these algorithms often require the computation of elementary functions involving such operations as vector rotations and exponentials. The CORDIC (COordinate Rotation Digital Computer) algorithms are the natural candidates for these elementary operations and they lead to novel realizations in many promising algorithms such as the square-root normalized ladder algorithms. FFT and several well-known matrix algorithms for signal processing. The development of algorithms involving nonstationary processes and the limiting constraints of VLSI motivate the extension of the CORDIC concept in other directions. It can be applied not only to one-parameter groups but to other Lie groups as well, i.e., higher dimensional manifolds, leading to generalized CORDIC algorithms. The computation of special functions is some of the potential benefits. Several such generalizations will be discussed. © 1982 IEEE.

关键词： Computational efficiency

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