Multistage adaptive filtering in a multirate digital signal processing system

Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1993.Includes bibliographical references (leaves 101-104).by Jen Mei Chen.Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1993

Quantization effects in the polyphase N-path IIR structure

Polyphase IIR structures have recently proven themselves very attractive for very high performance filters that can be designed using very few coefficients. This, combined with their low sensitivity to coefficient quantization in comparison to standard FIR and IIR structures, makes them very applicable for very fast filtering when implemented in fixed-point arithmetic. However, although the mathematical description is very simple, there exist a number of ways to implement such filters. In this paper, we take four of these different implementation structures, analyze the rounding noise originating from the limited arithmetic wordlength of the mathematical operators, and check the internal data growth within the structure. These analyses need to be done to ensure that the performance of the implementation matches the performance of the theoretical design. The theoretical approach that we present has been proven by the results of the fixed-point simulation done in Simulink and verified by an equivalent bit-true implementation in VHDL

Blind adaptive constrained reduced-rank parameter estimation based on constant modulus design for CDMA interference suppression

This paper proposes a multistage decomposition for blind adaptive parameter estimation in the Krylov subspace with the code-constrained constant modulus (CCM) design criterion. Based on constrained optimization of the constant modulus cost function and utilizing the Lanczos algorithm and Arnoldi-like iterations, a multistage decomposition is developed for blind parameter estimation. A family of computationally efficient blind adaptive reduced-rank stochastic gradient (SG) and recursive least squares (RLS) type algorithms along with an automatic rank selection procedure are also devised and evaluated against existing methods. An analysis of the convergence properties of the method is carried out and convergence conditions for the reduced-rank adaptive algorithms are established. Simulation results consider the application of the proposed techniques to the suppression of multiaccess and intersymbol interference in DS-CDMA systems

Synthesis methods for linear-phase FIR filters with a piecewise-polynomial impulse response

his thesis concentrates on synthesis methods for linear-phase finite-impulse response filters with a piecewise-polynomial impulse response. One of the objectives has been to find integer-valued coefficients to efficiently implement filters of the piecewise-polynomial impulse response approach introduced by Saram¨aki and Mitra. In this method, the impulse response is divided into blocks of equal length and each block is created by a polynomial of a given degree. The arithmetic complexity of these filters depends on the polynomial degree and the number of blocks. By using integer-valued coefficients it is possible to make the implementation of the subfilters, which generates the polynomials, multiplication-free. The main focus has been on finding computationally-efficient synthesis methods by using a piecewise-polynomial and a piecewise-polynomial-sinusoidal impulse responses to make it possible to implement high-speed, low-power, highly integrated digital signal processing systems. The earlier method by Chu and Burrus has been studied. The overall impulse response of the approach proposed in this thesis consists of the sum of several polynomial-form responses. The arithmetic complexity depends on the polynomial degree and the number of polynomial-form responses. The piecewise-polynomial-sinusoidal approach is a modification of the piecewise-polynomial approach. The subresponses are multiplied by a sinusoidal function and an arbitrary number of separate center coefficients is added. Thereby, the arithmetic complexity depends also on the number of complex multipliers and separately generated center coefficients. The filters proposed in this thesis are optimized by using linear programming methods

Modern VLSI Analogue Filter Design: Methodology and Software Development

This thesis describes various approaches for the design of modern analogue filters and provides a practical filter and equaliser design aids system XFILT. The thesis begins by placing the analogue filter design technique and software into a historical and technology perspective. The evolution of the analogue filter is traced from early work, through the passive-RLC to transconductor-C and switched-current realisations. The software development in VLSI analogue filter automation is reviewed. For SC filter design, a cascade SC design approach which includes a novel pole-zero pairing method and a comprehensive comparison of SC filter realisation using different biquads are presented. Very useful guidelines for the choice of a suitable biquad structure according to the nature of the filter problem are presented. The canonical realisations of SC filter are studied. The multirate SC system design is described. Several strategies and the algorithms for multirate SC system design are proposed. In transconductor-C filter design research, the definition of a canonical ladder based transconductor-C filter is introduced, and two canonical ladder based transconductor-C filter design approaches are proposed. The ladder based transconductor-C equaliser design is also discussed. A practical video frequency transconductor-C filter and equaliser design is given to demonstrate the utility of the matrix design method and the design software. A new approach to realise exact ladder based SI filter with first and second generation memory cell has been proposed. The bilinear transformation is used in the design procedure. Eight different SI ladder based structures can be obtained for one prototype ladder. Therefore it provides SI filter designers with various circuit choices based on different requirement such as area, maximum ratio of transistor aspect ratio limit, sensitivity or noise performance. Techniques to improve dynamic range and reduce circuit parameter spread are also presented. The proposed approach is well suited for a computer compiler implementation. A suitability study of each decomposition method for different filtering applications is also carried out and a general guideline for the choice of different decomposition methods is obtained. A comparison study on SI filter sensitivity performance based on first generation and second generation memory cells is carried out. Using four filter examples, it is demonstrated that SI filters based on a second generation SI memory cell have good sensitivity performance. For SI filters based on first generation memory cells, it is shown that a high ratio of clock frequency to cutoff frequency in the lowpass case, or a high ratio of clock frequency to midband frequency in the bandpass case would introduce high sensitivity. A novel approach for SI ladder filter based on the S2I integrator is also proposed and a canonical realisation for SI filter based on S2I integrator is developed. Examination of SI equaliser design reveals that cascade structure is a better candidate than ladder based structure. Multirate SI filter system design is also studied. Finally, a very brief introduction to the assembly of the design methods in this thesis into a software package XHLT for VLSI analogue filter and equaliser design is given. The user aspects of XFILT have been discussed and various capabilities of XFILT are demonstrated. Several advanced facilities which remove traditional design limitations are illustrated. The philosophy of the system is explained. It is shown that the distinguished features of XFILT are Ease of Use. General Applicability, and Ease of Extension. The system structure is described and the graphics interface which acts both as user friendly interface and a system manager of all the software is outlined. Fabricated SC, transconductor-C, and SI filter and equaliser have been designed by using XFILT. The system is under further enhancement toward a commercial product

Digital Prototype Filter Alternatives for Processing Frequency-Stacked Mobile Subbands Deploying a Single ADC for Beamforming Satellites

This article presents a two-stage approach for the processing of frequency-stacked mobile subbands. The frequency stacking is performed in the analog domain to enable the use of a wideband analog-to-digital converter (ADC), instead of employing multiple narrowband ADCs, to support multiple antenna elements for digital satellite beamforming. This analog front end provides a common broadband digital interface to the on-board processor and can be configured to support multiple satellite missions, reducing the cost of commissioning a digital processor for individual satellite missions. This article proposes a framework on the specification of digital prototype filter for the analysis of frequency-stacked mobile subbands. The computational complexity of the analysis operation, with two digital filter alternatives, are evaluated. A series of results, taken from our European Space Agency sponsored project, are presented here to demonstrate the applicability of the proposed two stage approach, reporting on the savings in power consumption when an Nth-band all-pass-based recursive filter having an infinite impulse response is used as the digital prototype filter

Accurate sound synthesis of 3D object collisions in interactive virtual scenarios

Questa tesi affronta lo studio di algoritmi efficienti per la sintesi di suoni risultanti dalla collisione di oggetti generici, partendo da una descrizione fisica del problema. L'obiettivo della ricerca e' lo sviluppo di strumenti in grado di aumentare l'accuratezza del feedback uditivo in ambienti di realta' virtuale attraverso un approccio basato sulla fisica, senza il bisogno quindi di far riferimento a suoni pre-registrati. Data la loro versatilita' nel trattare geometrie complesse, i metodi agli elementi finiti (FEM) sono stati scelti per la discretizzazione spaziale di generici risonatori tridimensionali. Le risultanti equazioni discrete sono riarrangiate in modo da disaccoppiare i modi normali del sistema tramite l'utilizzo di tecniche di Analisi e Sintesi Modale. Queste tecniche, infatti, portano convenientemente ad algoritmi computazionalmente efficienti per la sintesi del suono. Implementazioni di esempio di tali algoritmi sono state sviluppate facendo uso solo di software open-source: questo materiale a corredo della tesi permette una migliore riproducibilita' dei risultati di questa tesi da parte di ricercatori aventi una preparazione nel campo della sintesi audio. I risultati originali presenti in questo lavoro includono: i tecniche efficienti basate sulla fisica che aiutano l'implementazione in tempo reale di algoritmi di sintesi del suono su hardware comune; ii un metodo per la gestione efficiente dei dati provenienti da analisi FEM che, assieme ad un modello espressivo per la dissipazione, permette di calcolare l'informazione caratterizzante un oggetto risonante e salvarla in una struttura dati compatta iii una trasformazione nel dominio discreto del tempo su due diverse rappresentazioni nello spazio degli stati di filtri digitali del secondo ordine, che permette il calcolo esatto di variabili derivate come la velocita' e l'energia di un risonatore anche quando semplici realizzazioni a soli poli sono impiegate i un'efficiente realizzazione multirate di un banco parallelo di risonatori, derivata usando una suddivisione con Quadrature-Mirror-Filters (QMF). Confrontata con lavori simili presenti in letteratura, questa realizzazione permette l'uso di eccitazione nonlineare in feedback per un banco di risonatori in multirate: l'idea chiave consiste nello svolgere un cambio di stato adattivo nel banco di risonatori, muovendo i risonatori dalla frequenza di campionamento elevata, usata per il processamento della fase transiente, ad un insieme di sottofrequenze ridotte usate durante l'evoluzione in stato libero del sistema.This thesis investigates efficient algorithms for the synthesis of sounds produced by colliding objects, starting from a physical description of the problem. The objective of this investigation is to provide tools capable of increasing the accuracy of the synthetic auditory feedback in virtual environments through a physics-based approach, hence without the need of pre-recorded sounds. Due to their versatility in dealing with complex geometries, Finite Element Methods (FEM) are chosen for the space-domain discretization of generic three-dimensional resonators. The resulting state-space representations are rearranged so as to decouple the normal modes in the corresponding equations, through the use of Modal Analysis/Synthesis techniques. Such techniques, in fact, conveniently lead to computationally efficient sound synthesis algorithms. The whole mathematical treatment develops until deriving such algorithms. Finally, implementation examples are provided which rely only on open-source software: this companion material guarantees the reproducibility of the results, and can be handled without much effort by most researchers having a background in sound processing. The original results presented in this work include: i efficient physics-based techniques that help implement real-time sound synthesis algorithms on common hardware; ii a method for the efficient management of FEM data which, by working together with an expressive damping model, allows to pre-compute the information characterizing a resonating object and then to store it in a compact data structure; iii a time-domain transformation of the state-space representation of second-order digital filters, allowing for the exact computation of dependent variables such as resonator velocity and energy, even when simple all-pole realizations are used; iv an efficient multirate realization of a parallel bank of resonators, which is derived using a Quadrature-Mirror-Filters (QMF) subdivision. Compared to similar works previously proposed in the literature, this realization allows for the nonlinear feedback excitation of a multirate filter bank: the key idea is to perform an adaptive state change in the resonator bank, by switching the sampling rate of the resonators from a common highest value, used while processing the initial transient of the signals at full bandwidth, to a set of lower values in ways to enable a multirate realization of the same bank during the steady state evolution of the signals

Channelization for Multi-Standard Software-Defined Radio Base Stations

As the number of radio standards increase and spectrum resources come under more pressure, it becomes ever less efficient to reserve bands of spectrum for exclusive use by a single radio standard. Therefore, this work focuses on channelization structures compatible with spectrum sharing among multiple wireless standards and dynamic spectrum allocation in particular. A channelizer extracts independent communication channels from a wideband signal, and is one of the most computationally expensive components in a communications receiver. This work specifically focuses on non-uniform channelizers suitable for multi-standard Software-Defined Radio (SDR) base stations in general and public mobile radio base stations in particular. A comprehensive evaluation of non-uniform channelizers (existing and developed during the course of this work) shows that parallel and recombined variants of the Generalised Discrete Fourier Transform Modulated Filter Bank (GDFT-FB) represent the best trade-off between computational load and flexibility for dynamic spectrum allocation. Nevertheless, for base station applications (with many channels) very high filter orders may be required, making the channelizers difficult to physically implement. To mitigate this problem, multi-stage filtering techniques are applied to the GDFT-FB. It is shown that these multi-stage designs can significantly reduce the filter orders and number of operations required by the GDFT-FB. An alternative approach, applying frequency response masking techniques to the GDFT-FB prototype filter design, leads to even bigger reductions in the number of coefficients, but computational load is only reduced for oversampled configurations and then not as much as for the multi-stage designs. Both techniques render the implementation of GDFT-FB based non-uniform channelizers more practical. Finally, channelization solutions for some real-world spectrum sharing use cases are developed before some final physical implementation issues are considered