Shuttle Ku-band and S-band communications implementations study

The interfaces between the Ku-band system and the TDRSS, between the S-band system and the TDRSS, GSTDN and SGLS networks, and between the S-band payload communication equipment and the other Orbiter avionic equipment were investigated. The principal activities reported are: (1) performance analysis of the payload narrowband bent-pipe through the Ku-band communication system; (2) performance evaluation of the TDRSS user constraints placed on the S-band and Ku-band communication systems; (3) assessment of the shuttle-unique S-band TDRSS ground station false lock susceptibility; (4) development of procedure to make S-band antenna measurements during orbital flight; (5) development of procedure to make RFI measurements during orbital flight to assess the performance degradation to the TDRSS S-band communication link; and (6) analysis of the payload interface integration problem areas

IUS/payload communication system simulator configuration definition study

The requirements and specifications for a general purpose payload communications system simulator to be used to emulate those communications system portions of NASA and DOD payloads/spacecraft that will in the future be carried into earth orbit by the shuttle are discussed. For the purpose of on-orbit checkout, the shuttle is required to communicate with the payloads while they are physically located within the shuttle bay (attached) and within a range of 20 miles from the shuttle after they have been deployed (detached). Many of the payloads are also under development (and many have yet to be defined), actual payload communication hardware will not be available within the time frame during which the avionic hardware tests will be conducted. Thus, a flexible payload communication system simulator is required

Digit-slicing architectures for real-time digital filters

One of the many important algorithmic techniques in digital signal processing is real-time digital filtering. Modular sliced structures for digital filters have been proposed before, but the nature of implementation has been mainly constrained to non-recursive second order digital filters with positive values of coefficients. The aim of this research project is to extend this modular digit slicing concept to more practical higher order digital filters which are recursive and are of many forms (direct, nondirect, canonic, non-canonic). [Continues.

Signal processing using short word-length

Recently short word-length (normally 1 bit or bits) processing has become a promising technique. However, there are unresolved issues in sigma-delta modulation, which is the basis for 1b/2b systems. These issues hindered the full adoption of single-bit techniues in industry. Among these problems is the stability of high-order modulators and the limit cycle behaviour. More importantly, there is no adaptive LMS structure of any kind in 1b/2b domain. The challenge in this problem is the harsh quantization that prevents straightforward LMS application. In this thesis, the focus has been made on three axes: designing new single-bit DSP applications, proposing novel approaches for stability analysis, and tacking the unresolved problems of 1b/2b adaptive filtering. Two structures for 1b digital comb filtering are proposed. A ternary DC blocker structure is also presented and performanc e is tested. We also proposed a single-bit multiplierless DC-blocking structure. The stability of a single-bit high-order signma-delta modulator is studied under dc inputs. A new approach for stability analysis is proposed based on analogy with PLL analysis. Finally we succeeded in designing 1b/2b Wiener-like filtering and introduced (for the first time) three 1b/2b adaptive schemes

Investigation into synchronization for partial response signals and the development of a clock recovery scheme for 49QPRS signals

ThesisData communication is used increasingly in modern society. It is against this background that research is conducted worldwide toward the improvement of existing, as well as the development of new, improved communication techniques. Correlative encoding of data before transmission IS a very frequency-effective communication technique. The extent to which any communication technique is used, however, is dependent on a wide variety of factors. This study regarding the synchronisation of 49QPRS signals was undertaken with this in mind. Since digital signal processing (DSP) is used increasingly in modern communication systems, both a data transmitter and receiver were implemented by making use of this technique. Not only would this result in a system with all the desirable characteristics inherent to DSP, but, by making limited changes to the supporting software, the evaluation of a wide variety of alternatives became feasible. During the study a system making use of a pilot tone at one third the frequency of the carrier frequency was developed. The receiver recovers this signal by means of DSP techniques and its frequency is tripled. The phase of this recovered signal is crosscorrelated every 650 ~s in time with a locally generated signal of the correct frequency - and the phase of the locally generated signal is adjusted accordingly. It was found that the accuracy and stability of the locally generated signal were such that sufficient synchronisation was obtained in this manner. The quality of synchronisation is a function of the level of the pilot tone and if this tone should decrease to below a certain value, unacceptably large phase adjustments have to be made. This results in a senous degradation of the spectral purity of the recovered signal. However, the system as described exhibits extremely good noise immunity. During the development of the clock frequency recovery system, a baseband filter with a unique frequency response was defined. Making use of this, in conjunction with a limited amount of pre-processing, and an absolute value rectifier, recovery of the clock frequency becomes possible. In order to limit the amount of processing by the receiver, the baseband filter was implemented in its entirety in the transmitter. The recovered signal showed a moderate amount of amplitude variation, but an extremely stable synchronising signal could be derived from this. During the study both levels of synchronisation required by a hypothetical 49QPRS data communication system were therefore investigated fully and solutions found

Window Functions and Their Applications in Signal Processing

Window functions—otherwise known as weighting functions, tapering functions, or apodization functions—are mathematical functions that are zero-valued outside the chosen interval. They are well established as a vital part of digital signal processing. Window Functions and their Applications in Signal Processing presents an exhaustive and detailed account of window functions and their applications in signal processing, focusing on the areas of digital spectral analysis, design of FIR filters, pulse compression radar, and speech signal processing. Comprehensively reviewing previous research and recent developments, this book: Provides suggestions on how to choose a window function for particular applications Discusses Fourier analysis techniques and pitfalls in the computation of the DFT Introduces window functions in the continuous-time and discrete-time domains Considers two implementation strategies of window functions in the time- and frequency domain Explores well-known applications of window functions in the fields of radar, sonar, biomedical signal analysis, audio processing, and synthetic aperture rada

Window Functions and Their Applications in Signal Processing

Window functions—otherwise known as weighting functions, tapering functions, or apodization functions—are mathematical functions that are zero-valued outside the chosen interval. They are well established as a vital part of digital signal processing. Window Functions and their Applications in Signal Processing presents an exhaustive and detailed account of window functions and their applications in signal processing, focusing on the areas of digital spectral analysis, design of FIR filters, pulse compression radar, and speech signal processing. Comprehensively reviewing previous research and recent developments, this book: Provides suggestions on how to choose a window function for particular applications Discusses Fourier analysis techniques and pitfalls in the computation of the DFT Introduces window functions in the continuous-time and discrete-time domains Considers two implementation strategies of window functions in the time- and frequency domain Explores well-known applications of window functions in the fields of radar, sonar, biomedical signal analysis, audio processing, and synthetic aperture rada

Engineering evaluations and studies. Volume 3: Exhibit C

High rate multiplexes asymmetry and jitter, data-dependent amplitude variations, and transition density are discussed

Design and analysis of short word length DSP systems for mobile communication

Recently, many general purpose DSP applications such as Least Mean Squares-Like single-bit adaptive filter algorithms have been developed using the Short Word Length (SWL) technique and have been shown to achieve similar performance as multi-bit systems. A key function in SWL systems is sigma delta modulation (ΣΔM) that operates at an over sampling ratio (OSR), in contrast to the Nyquist rate sampling typically used in conventional multi-bit systems. To date, the analysis of SWL (or single-bit) DSP systems has tended to be performed using high-level tools such as MATLAB, with little work reported relating to their hardware implementation, particularly in Field Programmable Gate Arrays (FPGAs). This thesis explores the hardware implementation of single-bit systems in FPGA using the design and implementation in VHDL of a single-bit ternary FIR-like filter as an illustrative example. The impact of varying OSR and bit-width of the SWL filter has been determined, and a comparison undertaken between the area-performance-power characteristics of the SWL FIR filter compared to its equivalent multi-bit filter. In these experiments, it was found that single-bit FIR-like filter consistently outperforms the multi-bit technique in terms of its area, performance and power except at the highest filter orders analysed in this work. At higher orders, the ΣΔ approach retains its power and performance advantages but exhibits slightly higher chip area. In the second stage of thesis, three encoding techniques called canonical signed digit (CSD), 2’s complement, and Redundant Binary Signed Digit (RBSD) were designed and investigated on the basis of area-performance in FPGA at varying OSR. Simulation results show that CSD encoding technique does not offer any significant improvement as compared to 2’s complement as in multi-bit domain. Whereas, RBSD occupies double the chip area than other two techniques and has poor performance. The stability of the single-bit FIR-like filter mainly depends upon IIR remodulator due to its recursive nature. Thus, we have investigated the stability IIR remodulator and propose a new model using linear analysis and root locus approach that takes into account the widely accepted second order sigma-delta modulator state variable upper bounds. Using proposed model we have found new feedback parameters limits that is a key parameter in single-bit IIR remodulator stability analysis. Further, an analysis of single-bit adaptive channel equalization in MATLAB has been performed, which is intended to support the design and development of efficient algorithm for single-bit channel equalization. A new mathematical model has been derived with all inputs, coefficients and outputs in single-bit domain. The model was simulated using narrowband signals in MATLAB and investigated on the basis of symbol error rate (SER), signal-to-noise ratio (SNR) and minimum mean squared error (MMSE). The results indicate that single-bit adaptive channel equalization is achievable with narrowband signals but that the harsh quantization noise has great impact in the convergence