The Generalised Instrument

Thesis (M.E.Sc.) -- University of Adelaide, 199

Design of software radio

Software Define Radio (SDR) has become a prevalent technology in wireless systems. In SDR some or all of the signal specific handling is implemented in software functions, while other functions like decimation, interpolation, digital up-conversion and digital down conversion are done on reprogrammable Digital Signal Processor or Field Programmable Gate Arrays.Twelve laboratory exercises have been designed to lead the student through the process of using the Universal Software Radio peripheral (USRP) hardware and GNU Radio open source software

Design of software radio

Software Define Radio (SDR) has become a prevalent technology in wireless systems. In SDR some or all of the signal specific handling is implemented in software functions, while other functions like decimation, interpolation, digital up-conversion and digital down conversion are done on reprogrammable Digital Signal Processor or Field Programmable Gate Arrays.Twelve laboratory exercises have been designed to lead the student through the process of using the Universal Software Radio peripheral (USRP) hardware and GNU Radio open source software

34th Midwest Symposium on Circuits and Systems-Final Program

Organized by the Naval Postgraduate School Monterey California. Cosponsored by the IEEE Circuits and Systems Society. Symposium Organizing Committee: General Chairman-Sherif Michael, Technical Program-Roberto Cristi, Publications-Michael Soderstrand, Special Sessions- Charles W. Therrien, Publicity: Jeffrey Burl, Finance: Ralph Hippenstiel, and Local Arrangements: Barbara Cristi

A generalized, parametric PR-QMF/wavelet transform design approach for multiresolution signal decomposition

This dissertation aims to emphasize the interrelations and the linkages of the theories of discrete-time filter banks and wavelet transforms. It is shown that the Binomial-QMF banks are identical to the interscale coefficients or filters of the compactly supported orthonormal wavelet transform bases proposed by Daubechies. A generalized, parametric, smooth 2-band PR-QMF design approach based on Bernstein polynomial approximation is developed. It is found that the most regular compact support orthonormal wavelet filters, coiflet filters are only the special cases of the proposed filter bank design technique. A new objective performance measure called Non-aliasing Energy Ratio(NER) is developed. Its merits are proven with the comparative performance studies of the well known orthonormal signal decomposition techniques. This dissertation also addresses the optimal 2-band PR-QMF design problem. The variables of practical significance in image processing and coding are included in the optimization problem. The upper performance bounds of 2-band PR-QMF and their corresponding filter coefficients are derived. It is objectively shown that there are superior filter bank solutions available over the standard block transform, DCT. It is expected that the theoretical contributions of this dissertation will find its applications particularly in Visual Signal Processing and Coding

Applications of fuzzy counterpropagation neural networks to non-linear function approximation and background noise elimination

An adaptive filter which can operate in an unknown environment by performing a learning mechanism that is suitable for the speech enhancement process. This research develops a novel ANN model which incorporates the fuzzy set approach and which can perform a non-linear function approximation. The model is used as the basic structure of an adaptive filter. The learning capability of ANN is expected to be able to reduce the development time and cost of the designing adaptive filters based on fuzzy set approach. A combination of both techniques may result in a learnable system that can tackle the vagueness problem of a changing environment where the adaptive filter operates. This proposed model is called Fuzzy Counterpropagation Network (Fuzzy CPN). It has fast learning capability and self-growing structure. This model is applied to non-linear function approximation, chaotic time series prediction and background noise elimination

Digital signal conditioning on multiprocessor systems

An important application area of modem computer systems is that of digital signal processing. This discipline is concerned with the analysis or modification of digitally represented signals, through the use of simple mathematical operations. A primary need of such systems is that of high data throughput. Although optimised programmable processors are available, system designers are now looking towards parallel processing to gain further performance increases. Such parallel systems may be easily constructed using the transputer family of processors. However, although these devices are comparatively easy to program, they possess a general von Neumann core and so are relatively inefficient at implementing digital signal processing algorithms. The power of the transputer lies in its ability to communicate effectively, not in its computational capability. The converse is true of specialised digital signal processors. These devices have been designed specifically to implement the type of small data intensive operations required by digital signal processing algorithms, but have not been designed to operate efficiently in a multiprocessor environment. This thesis examines the performance of both types of processors with reference to a common signal processing application, multichannel filtering. The transputer is examined in both uniprocessor and multiprocessor configurations, and its performance analysed. A theoretical model of program behaviour is developed, in order to assess the performance benefits of particular code structures and the effects of such parameters as data block size. The transputer implementation is contrasted with that of the Motorola DSP56001 digital signal processor. This device is found to be much more efficient at implementing such algorithms on a single device, but provides limited multiprocessor support. Using the conclusions of this assessment, a hybrid multiprocessor has been designed. This consists of a transputer controlling a number of signal processors, communicating through shared memory, separating tiie tasks of computation and communication. Forcing the transputer to communicate through shared memory causes problems, and these have been addressed. A theoretical performance model of the system has been produced. A small system has been constructed, and is currently running performance test software

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

Proceedings of the Mobile Satellite Conference

A satellite-based mobile communications system provides voice and data communications to mobile users over a vast geographic area. The technical and service characteristics of mobile satellite systems (MSSs) are presented and form an in-depth view of the current MSS status at the system and subsystem levels. Major emphasis is placed on developments, current and future, in the following critical MSS technology areas: vehicle antennas, networking, modulation and coding, speech compression, channel characterization, space segment technology and MSS experiments. Also, the mobile satellite communications needs of government agencies are addressed, as is the MSS potential to fulfill them

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook