Blind adaptive equalization for QAM signals: New algorithms and FPGA implementation.

Adaptive equalizers remove signal distortion attributed to intersymbol interference in band-limited channels. The tap coefficients of adaptive equalizers are time-varying and can be adapted using several methods. When these do not include the transmission of a training sequence, it is referred to as blind equalization. The radius-adjusted approach is a method to achieve blind equalizer tap adaptation based on the equalizer output radius for quadrature amplitude modulation (QAM) signals. Static circular contours are defined around an estimated symbol in a QAM constellation, which create regions that correspond to fixed step sizes and weighting factors. The equalizer tap adjustment consists of a linearly weighted sum of adaptation criteria that is scaled by a variable step size. This approach is the basis of two new algorithms: the radius-adjusted modified multitmodulus algorithm (RMMA) and the radius-adjusted multimodulus decision-directed algorithm (RMDA). An extension of the radius-adjusted approach is the selective update method, which is a computationally-efficient method for equalization. The selective update method employs a stop-and-go strategy based on the equalizer output radius to selectively update the equalizer tap coefficients, thereby, reducing the number of computations in steady-state operation. (Abstract shortened by UMI.) Source: Masters Abstracts International, Volume: 45-01, page: 0401. Thesis (M.A.Sc.)--University of Windsor (Canada), 2006

Advanced Modulation and Coding Technology Conference

The objectives, approach, and status of all current LeRC-sponsored industry contracts and university grants are presented. The following topics are covered: (1) the LeRC Space Communications Program, and Advanced Modulation and Coding Projects; (2) the status of four contracts for development of proof-of-concept modems; (3) modulation and coding work done under three university grants, two small business innovation research contracts, and two demonstration model hardware development contracts; and (4) technology needs and opportunities for future missions

Strategies for Devising Automatic Signal Recognition Algorithms in a Shared Radio Environment

In an increasingly congested and complex radio environment interference is to be expected, which poses problems for Automatic Signal Recognition (ASR) systems. This thesis explores strategies for improving ASR performance in the presence of interference. The thesis breaks the overall research question down into a number of subquestions and explores each of these in turn. A Phase-symmetric Cross Recurrence Plot is developed and used to show how a radio signal can be manipulated to separate information about the modulation from the information being carried. The Logarithmic Cyclic frequency Domain Profile is introduced to illustrate how a logarithmic representation can be used for analysing mixtures of signals with very different cyclic frequencies. After defining a canonical ASR system architecture, the concepts of an Ideal Feature and Interference Selectivity are introduced and applied to typical features used in ASR processing. Finally it is shown how these algorithmic developments can be combined in a Bayesian chain implementation that can accommodate a wide variety of feature extraction algorithms. It is concluded that future ASR systems will require features that can handle a wide range of signal types with much higher levels of interference selectivity if they are to achieve acceptable performance in shared spectrum bands. Intelligent segmentation is shown to be a requirement for future ASR systems unless features can be developed that have near ideal performance

Direct digital synthesizers : theory, design and applications

Traditional designs of high bandwidth frequency synthesizers employ the use of a phase-locked-loop (PLL). A direct digital synthesizer (DDS) provides many significant advantages over the PLL approaches. Fast settling time, sub-Hertz frequency resolution, continuous-phase switching response and low phase noise are features easily obtainable in the DDS systems. Although the principle of the DDS has been known for many years, the DDS did not play a dominant role in wideband frequency generation until recent years. Earlier DDSs were limited to produce narrow bands of closely spaced frequencies, due to limitations of digital logic and D/A-converter technologies. Recent advantages in integrated circuit (IC) technologies have brought about remarkable progress in this area. By programming the DDS, adaptive channel bandwidths, modulation formats, frequency hopping and data rates are easily achieved. This is an important step towards a "software-radio" which can be used in various systems. The DDS could be applied in the modulator or demodulator in the communication systems. The applications of DDS are restricted to the modulator in the base station. The aim of this research was to find an optimal front-end for a transmitter by focusing on the circuit implementations of the DDS, but the research also includes the interface to baseband circuitry and system level design aspects of digital communication systems. The theoretical analysis gives an overview of the functioning of DDS, especially with respect to noise and spurs. Different spur reduction techniques are studied in detail. Four ICs, which were the circuit implementations of the DDS, were designed. One programmable logic device implementation of the CORDIC based quadrature amplitude modulation (QAM) modulator was designed with a separate D/A converter IC. For the realization of these designs some new building blocks, e.g. a new tunable error feedback structure and a novel and more cost-effective digital power ramp generator, were developed.reviewe

Adaptive equalisation for fading digital communication channels

This thesis considers the design of new adaptive equalisers for fading digital communication channels. The role of equalisation is discussed in the context of the functions of a digital radio communication system and both conventional and more recent novel equaliser designs are described. The application of recurrent neural networks to the problem of equalisation is developed from a theoretical study of a single node structure to the design of multinode structures. These neural networks are shown to cancel intersymbol interference in a manner mimicking conventional techniques and simulations demonstrate their sensitivity to symbol estimation errors. In addition the error mechanisms of conventional maximum likelihood equalisers operating on rapidly time-varying channels are investigated and highlight the problems of channel estimation using delayed and often incorrect symbol estimates. The relative sensitivity of Bayesian equalisation techniques to errors in the channel estimate is studied and demonstrates that the structure's equalisation capability is also susceptible to such errors. Applications of multiple channel estimator methods are developed, leading to reduced complexity structures which trade performance for a smaller computational load. These novel structures are shown to provide an improvement over the conventional techniques, especially for rapidly time-varying channels, by reducing the time delay in the channel estimation process. Finally, the use of confidence measures of the equaliser's symbol estimates in order to improve channel estimation is studied and isolates the critical areas in the development of the technique — the production of reliable confidence measures by the equalisers and the statistics of symbol estimation error bursts

Data transmission oriented on the object, communication media, application, and state of communication systems tactical communication system application

A proposed communication system architecture is denoted TOMAS, which stands for data Transmission oriented on the Object, communication Media, Application, and state of communication Systems. Given particular tactical communication system scenarios of image transmission over a wireless LOS (Line-of-Sight) channel, a wireless TOMAS system demonstrates superior performance compared to the conventional system, which is a combination of JPEG2000 image compression and OFDM transmission, in restored image quality parameters over a wide range of wireless channel parameters. The wireless TOMAS system provides progressive lossless image transmission under the influence of moderate fading without any kind of channel coding and estimation. The TOMAS system employs a fast proprietary patent pending algorithm Sabelkin (2011), which does not employ any multiplications, and it uses three times less real additions than the algorithm of JPEG2000+OFDM. The TOMAS system exploits a specialized wavelet transform combined for image coding and channel modulation