The rapid growth of technology in recent decades has changed the whole dimension of communications. Today people are more interested in hands-free communication. In such a situation, the use a regular loudspeaker and a high-gain microphone, in place of a telephone receiver, might seem more appropriate. This would allow more than one person to participate in a conversation at the same time such as a teleconference environment. Another advantage is that it would allow the person to have both hands free and to move freely in the room. However, the presence of a large acoustic coupling between the loudspeaker and microphone would produce a loud echo that would make conversation difficult. Furthermore, the acoustic system could become instable, which would produce a loud howling noise to occur. The solution to these problems is the elimination of the echo with an echo suppression or echo cancellation algorithm. The echo suppressor offers a simple but effective method to counter the echo problem. However, the echo suppressor possesses a main disadvantage since it supports only half-duplex communication. Half-duplex communication permits only one speaker to talk at a time. This drawback led to the invention of echo cancellers. An important aspect of echo cancellers is that full-duplex communication can be maintained, which allows both speakers to talk at the same time. This objective of this research was to produce an improved echo cancellation algorithm, which is capable of providing convincing results. The three basic components of an echo canceller are an adaptive filter, a doubletalk detector and a nonlinear processor. The adaptive filter creates a replica of the echo and subtracts it from the combination of the actual echo and the near-end signal. The doubletalk detector senses the doubletalk. Doubletalk occurs when both ends are talking, which stops the adaptive filter in order to avoid divergence. Finally, the nonlinear processor removes the residual echo from the error signal. Usually, a certain amount of speech is clipped in the final stage of nonlinear processing. In order to avoid clipping, a noise gate was used as a nonlinear processor in this research. The noise gate allowed a threshold value to be set and all signals below the threshold were removed. This action ensured that only residual echoes were removed in the final stage. To date, the real time implementation of echo an cancellation algorithm was performed by utilizing both a VLSI processor and a DSP processor. Since there has been a revolution in the field of personal computers, in recent years, this research attempted to implement the acoustic echo canceller algorithm on a natively running PC with the help of the MATLAB software.

<p>This master thesis consists of implementation and evaluation of an AEC, Acoustic Echo Canceller, algorithm in a floating-point architecture. The most important question this thesis will try to answer is to determine benefits or drawbacks of using a floating-point architecture, relative a fixed-point architecture, to do AEC. In a telephony system there is two common forms of echo, line echo and acoustic echo. Acoustic echo is introduced by sound emanating from a loudspeaker, e.g. in a handsfree or speakerphone, being picked up by a microphone and then sent back to the source. The problem with this feedback is that the far-end speaker will hear one, or multiple, time-delayed version(s) of her own speech. This time-delayed version of speech is usually perceived as both confusing and annoying unless removed by the use of AEC. In this master thesis the performance of a floating-point version of a normalized least-mean-square AEC algorithm was evaluated in an environment designed and implemented to approximate live telephony calls. An instruction-set simulator and assembler available at the initiation of this master thesis were extended to enable; zero-overhead loops, modular addressing, post-increment of registers and register-write forwarding. With these improvements a bit-true assembly version was implemented capable of real-time AEC requiring 15 million instructions per second. A solution using as few as eight mantissa bits, in an external format used when storing data in memory, was found to have an insignificant effect on the selected AEC implementation’s performance. Due to the relatively low memory requirement of the selected AEC algorithm, the use of a small external format has a minor effect on the required memory size. In total this indicates that the possible reduction of the memory requirement and related energy consumption, does not justify the added complexity and energy consumption of using a floating-point architecture for the selected algorithm. Use of a floating-point format can still be advantageous in speech-related signal processing when the introduced time delay by a subband, or a similar frequency domain, solution is unacceptable. Speech algorithms that have high memory use and small introduced delay requirements are a good candidate for a floating-point digital signal processor architecture.</p>

New adaptive-filtering algorithms, also known as adaptation algorithms, are proposed. The new algorithms can be broadly classified into two categories, namely, steepest-descent and Newton-type adaptation algorithms. Several new methods have been used to bring about improvements regarding the speed of convergence, steady-state misalignment, robustness with respect to impulsive noise, re-adaptation capability, and computational load of the proposed algorithms. In chapters 2, 3, and 8, several adaptation algorithms are developed that belong to the steepest-descent family. The algorithms of chapters 2 and 3 use two error bounds with the aim of reducing the computational load, achieving robust performance with respect to impulsive noise, good tracking capability and significantly reduced steady-state misalignment. The error bounds can be either prespecified or estimated using an update formula that incorporates a modified variance estimator. Analyses pertaining to the steady-state mean-square error (MSE) of some of these algorithms are also presented. The algorithms in chapter 8 use a so-called \textit{iterative/shrinkage method} to obtain a variable step size by which improved convergence characteristics can be achieved compared to those in other state-of-the-art competing algorithms. Several adaptation algorithms that belong to the Newton family are developed in chapters 4-6 with the aim of achieving robust performance with respect to impulsive noise, reduced steady-state misalignment, and good tracking capability without compromising the initial speed of convergence. The algorithm in chapter 4 imposes a bound on the $L_1$ norm of the gain vector in the crosscorrelation update formula to achieve robust performance with respect to impulsive noise in stationary environments. In addition to that, a variable forgetting factor is also used to achieve good tracking performance for applications in nonstationary environments. The algorithm in chapter 5 is developed to achieve a reduced steady-state misalignment and improved convergence speed and a reduced computational load. The algorithm in chapter 6 is essentially an extension of the algorithm in chapter 5 designed to achieve robust performance with respect to impulsive noise and reduced computational load. Analyses concerning the asymptotic stability and steady-state MSE of these algorithms are also presented. An algorithm that minimizes Reny's entropy of the error signal is developed in chapter 7 with the aim of achieving faster convergence and reduced steady-state misalignment compared to those in other algorithms of this family. Simulation results are presented that demonstrate the superior convergence characteristics of the proposed algorithms with respect to state-of-the-art competing algorithms of the same family in network-echo cancelation, acoustic-echo cancelation, system-identification, interference-cancelation, time-series prediction, and time-series filtering applications. In addition, simulation results concerning system-identification applications are also used to verify the accuracy of the MSE analyses presented. / Graduate

This master thesis consists of implementation and evaluation of an AEC, Acoustic Echo Canceller, algorithm in a floating-point architecture. The most important question this thesis will try to answer is to determine benefits or drawbacks of using a floating-point architecture, relative a fixed-point architecture, to do AEC. In a telephony system there is two common forms of echo, line echo and acoustic echo. Acoustic echo is introduced by sound emanating from a loudspeaker, e.g. in a handsfree or speakerphone, being picked up by a microphone and then sent back to the source. The problem with this feedback is that the far-end speaker will hear one, or multiple, time-delayed version(s) of her own speech. This time-delayed version of speech is usually perceived as both confusing and annoying unless removed by the use of AEC. In this master thesis the performance of a floating-point version of a normalized least-mean-square AEC algorithm was evaluated in an environment designed and implemented to approximate live telephony calls. An instruction-set simulator and assembler available at the initiation of this master thesis were extended to enable; zero-overhead loops, modular addressing, post-increment of registers and register-write forwarding. With these improvements a bit-true assembly version was implemented capable of real-time AEC requiring 15 million instructions per second. A solution using as few as eight mantissa bits, in an external format used when storing data in memory, was found to have an insignificant effect on the selected AEC implementation’s performance. Due to the relatively low memory requirement of the selected AEC algorithm, the use of a small external format has a minor effect on the required memory size. In total this indicates that the possible reduction of the memory requirement and related energy consumption, does not justify the added complexity and energy consumption of using a floating-point architecture for the selected algorithm. Use of a floating-point format can still be advantageous in speech-related signal processing when the introduced time delay by a subband, or a similar frequency domain, solution is unacceptable. Speech algorithms that have high memory use and small introduced delay requirements are a good candidate for a floating-point digital signal processor architecture.

Title from PDF of title page. / Document formatted into pages; contains 66 pages. / Thesis (M.S.E.E.)--University of South Florida, 2003. / Includes bibliographical references. / Text (Electronic thesis) in PDF format. / ABSTRACT: The rapid growth of technology in recent decades has changed the whole dimension of communications. Today people are more interested in hands-free communication. In such a situation, the use a regular loudspeaker and a high-gain microphone, in place of a telephone receiver, might seem more appropriate. This would allow more than one person to participate in a conversation at the same time such as a teleconference environment. Another advantage is that it would allow the person to have both hands free and to move freely in the room. However, the presence of a large acoustic coupling between the loudspeaker and microphone would produce a loud echo that would make conversation difficult. Furthermore, the acoustic system could become instable, which would produce a loud howling noise to occur. The solution to these problems is the elimination of the echo with an echo suppression or echo cancellation algorithm. / ABSTRACT: The echo suppressor offers a simple but effective method to counter the echo problem. However, the echo suppressor possesses a main disadvantage since it supports only half-duplex communication. Half-duplex communication permits only one speaker to talk at a time. This drawback led to the invention of echo cancellers. An important aspect of echo cancellers is that full-duplex communication can be maintained, which allows both speakers to talk at the same time. This objective of this research was to produce an improved echo cancellation algorithm, which is capable of providing convincing results. The three basic components of an echo canceller are an adaptive filter, a doubletalk detector and a nonlinear processor. The adaptive filter creates a replica of the echo and subtracts it from the combination of the actual echo and the near-end signal. The doubletalk detector senses the doubletalk. / ABSTRACT: Doubletalk occurs when both ends are talking, which stops the adaptive filter in order to avoid divergence. Finally, the nonlinear processor removes the residual echo from the error signal. Usually, a certain amount of speech is clipped in the final stage of nonlinear processing. In order to avoid clipping, a noise gate was used as a nonlinear processor in this research. The noise gate allowed a threshold value to be set and all signals below the threshold were removed. This action ensured that only residual echoes were removed in the final stage. To date, the real time implementation of echo an cancellation algorithm was performed by utilizing both a VLSI processor and a DSP processor. Since there has been a revolution in the field of personal computers, in recent years, this research attempted to implement the acoustic echo canceller algorithm on a natively running PC with the help of the MATLAB software. / System requirements: World Wide Web browser and PDF reader. / Mode of access: World Wide Web.

The least mean square (LMS) algorithm is widely expected to operate near the corresponding Wiener filter solution. An exception to this popular perception occurs when the algorithm is used to adapt a transversal equalizer in the presence of additive narrowband interference. The steady-state LMS equalizer behavior does not correspond to that of the fixed Wiener equalizer: the mean of its weights is different from the Wiener weights, and its mean squared error (MSE) performance may be significantly better than the Wiener performance. The contributions of this study serve to better understand this so-called non-Wiener phenomenon of the LMS and normalized LMS adaptive transversal equalizers. The first contribution is the analysis of the mean of the LMS weights in steady state, assuming a large interference-to-signal ratio (ISR). The analysis is based on the Butterweck expansion of the weight update equation. The equalization problem is transformed to an equivalent interference estimation problem to make the analysis of the Butterweck expansion tractable. The analytical results are valid for all step-sizes. Simulation results are included to support the analytical results and show that the analytical results predict the simulation results very well, over a wide range of ISR. The second contribution is the new MSE estimator based on the expression for the mean of the LMS equalizer weight vector. The new estimator shows vast improvement over the Reuter-Zeidler MSE estimator. For the development of the new MSE estimator, the transfer function approximation of the LMS algorithm is generalized for the steady-state analysis of the LMS algorithm. This generalization also revealed the cause of the breakdown of the MSE estimators when the interference is not strong, as the assumption that the variation of the weight vector around its mean is small relative to the mean of the weight vector itself. Both the expression for the mean of the weight vector and for the MSE estimator are analyzed for the LMS algorithm at first. The results are then extended to the normalized LMS algorithm by the simple means of adaptation step-size redefinition. / Ph. D.

Porttelefoner används ofta i högljudda miljöer. Ett exempel på en sådan miljö är vindutsatta områden, där operatören i ett rum kan uppleva svårigheter att uppfatta tal från användaren som talar i en porttelefon på grund av den omgivande höga ljudnivån. Många porttelefoner och andra liknande enheter stöter vanligtvis på utmaningar och begränsningar, särskilt när det gäller snabbhet, storlek, resurshantering och hantering av dynamiska signaler.  Detta projekt genomfördes vid ett svenskt företag inom nätverksbaserade lösningar för videoövervakning och fysisk säkerhet. Projektet syftar till att utforska och implementera ett adaptiv filter med en adaptiv algoritm i C-programmering för att komplettera ett digitalt signalbehandlingssystem som en strategi för att förbättra ljudkvaliteten genom att reducera bruset hos porttelefoner i utmanande miljöer. Genom att tillämpa ett lämpligt adaptiv filter i en Raspberry Pi för att simulera en porttelefon, strävar projektet efter att reducera brus och optimera talet. Några av de vanligaste filtreringsalgoritmerna som använts i tidigare forskning för att förbättra ljudkvaliteten är Least mean square, Normalized least mean square och Recursive least square som även utvärderas i denna studie. Efter noggranna studier valdes algoritmen Normalized least mean square för implementering i detta projekt. Algoritmens prestanda utvärderas med hjälp av beräkningstiden, medelkvadratfelet och signal-till-brus-förhållandet i Matlab samt användartester för att säkerställa kvaliteten. Detta projekt uppnådde målen genom att utveckla ett fungerande adaptivt filter. Det rekommenderas att implementera filtret i en porttelefon där mikrofonerna inte är placerade nära varandra för att förhindra upptagning av dubbla liknande signaler. Under projektets gång hanterade systemet kontinuerligt dataströmmar effektivt i praktiska tester, vilket bekräftade att det fungerade utan fördröjningar. Detta bevisade det adaptiva filtrets effektivitet i verkliga applikationer, särskilt i högljudda miljöer. / Intercoms are often used in noisy environments. An example of such an environment is windy areas, where the operator inside a room may find it difficult to perceive speech from a user speaking through an intercom due to the surrounding high noise levels. Many intercoms and other similar devices typically encounter challenges and limitations, especially in terms of speed, size, resource management, and handling of dynamic signals. This project was carried out at a Swedish company specializing in network-based solutions for video surveillance and physical security. The project’s objective was to study and implement an adaptive filter with an adaptive algorithm in C programming to complement a digital signal processing system, as a strategy to enhance sound quality by reducing noise in intercoms in challenging environments. By applying a suitable adaptive filter in a Raspberry Pi to simulate an intercom, the goal of the project is to reduce noise and optimize speech clarity. Some of the most common filtering algorithms used in previous research to improve sound quality include Least mean square, Normalized least mean square och Recursive least square, which are evaluated in this study. After thorough studies, the Normalized least mean square algorithm was selected for implementation in this project. The performance of the algorithm is assessed using computation time, mean squared error, and signal-to-noise ratio in Matlab, along with user testing to ensure quality. This project achieved its goals by developing a functional adaptive filter. It is recommended to implement the filter in an intercom where the microphones are not placed close to each other to prevent the capture of similar duplicate signals. Throughout the project, the system continuously handled data streams effectively in practical tests, confirming that it operated without delays. This demonstrated the adaptive filter's effectiveness in real applications, particularly in noisy environments.

Submitted by Cristiane Chim (cristiane.chim@ucpel.edu.br) on 2018-05-08T17:34:22Z No. of bitstreams: 1 Vagner Guidotti Furtado (1).pdf: 2942442 bytes, checksum: a43c18ecb28456284d4b6c622f11210d (MD5) / Made available in DSpace on 2018-05-08T17:34:22Z (GMT). No. of bitstreams: 1 Vagner Guidotti Furtado (1).pdf: 2942442 bytes, checksum: a43c18ecb28456284d4b6c622f11210d (MD5) Previous issue date: 2017-12-07 / The division operation in digital systems has its relevance because it is a necessary function in several applications, such as general purpose processors, digital signal processors and microcontrollers. The digital divider circuit is of great architectural complexity and may occupy a considerable area in the design of an integrated circuit, and as a consequence may have a great influence on the static and dynamic power dissipation of the circuit as a whole. In relation to the application of dividing circuits in circuits of the Digital Signal Processing (DSP) area, adaptive filters have a particular appeal, especially when using algorithms that perform a normalization in the input signals. In view of the above, this work focuses on the proposition of algorithms, techniques for reducing energy consumption and logical area, proposition and implementation of efficient dividing circuit architectures for use in adaptive filters. The Newton-Raphson and Goldschmidt iterative dividing circuits both operating at fixed-point were specifically addressed. The results of the synthesis of the implemented architectures of the divisors with the proposed algorithms and techniques showed considerable reduction of power and logical area of the circuits. In particular, the dividing circuits were applied in adaptive filter architectures based on the NLMS (Normalized least Mean Square) algorithm, seeking to add to these filters, characteristics of good convergence speed, combined with the improvement in energy efficiency. The adaptive filters implemented are used in the case study of harmonic cancellation on electrocardiogram signals / A operação de divisão em sistemas digitais tem sua relevância por se tratar de uma função necessária em diversas aplicações, tais como processadores de propósito geral, processadores digitais de sinais e microcontroladores. O circuito divisor digital é de grande complexidade arquitetural, podendo ocupar uma área considerável no projeto de um circuito integrado, e por consequência pode ter uma grande influência na dissipação de potência estática e dinâmica do circuito como um todo. Em relação à aplicação de circuitos divisores em circuitos da área DSP (Digital Signal Processing), os filtros adaptativos têm um particular apelo, principalmente quando são utilizados algoritmos que realizam uma normalização nos sinais de entrada. Diante do exposto, este trabalho foca na proposição de algoritmos, técnicas de redução de consumo de energia e área lógica, proposição e implementação de arquiteturas de circuitos divisores eficientes para utilização em filtros adaptativos. Foram abordados em específico os circuitos divisores iterativos Newton-Raphson e Goldschmidt ambos operando em ponto-fixo. Os resultados da síntese das arquiteturas implementadas dos divisores com os algoritmos e técnicas propostas mostraram considerável redução de potência e área lógica dos circuitos. Em particular, os circuitos divisores foram aplicados em arquiteturas de filtros adaptativos baseadas no algoritmo NLMS (Normalized least Mean Square), buscando agregar a esses filtros, características de boa velocidade de convergência, aliada à melhoria na eficiência energética. Os filtros adaptativos implementados são utilizados no estudo de caso de cancelamento de harmônicas em sinais de eletrocardiograma (ECG)

Made available in DSpace on 2016-08-17T14:52:41Z (GMT). No. of bitstreams: 1 Dissertacao-WilanderTestonePereiraSilva.pdf: 3903191 bytes, checksum: b91ff906a27937df64d75b330c6ea137 (MD5) Previous issue date: 2016-03-11 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / In this dissertation we present a research in aspects of stochastic modeling, convergence and applications of least mean square (LMS) algorithm, normalized least mean square (NLMS) algorithm and proportionate normalized least mean square (PNLMS) algorithm. Specifically, the aim is to address the LMS algorithm in your extension, defining his concepts, demonstrations of properties, algorithms and analysis of convergence, Learning Curve and Misadjustment of the algorithm in question. Within of the context of sensor networks and spatial filtering is evaluated the performance of the algorithms by the learning curve of the referred algorithms for arrangements of adaptive antennas. In the intrinsic context of the application in electrical engineering, in area of telecommunications that seek the best alternative and aims to optimize the process of transmission/reception to eliminate interference, and the least amount of elements in adaptive antenna arrays, which they are known as smart antenna, which aims to reach a signal noise ratio for small value, with appropriate number of elements. The performance of the LMS algorithm is evaluated in sensor networks that is characterized by an antenna array. Results of computer simulations for different scenarios of operation show that the algorithms have good numerical results of convergence to a suitable choice of the parameters related with the rate of learning that are associated with their average curves and the beamforming of the smart antenna array. / Nesta dissertação de mestrado apresenta-se uma investigação em aspectos de modelagem estocástica, convergência e aplicações dos algoritmos de mínimos quadrados médio (LMS), mínimos quadrados médio normalizado (NLMS) e mínimos quadrados médio normalizado proporcional (PNLMS). Particularmente, aborda-se o Algoritmo LMS em sua extensão, definindo conceitos, demonstrações de propriedades, algoritmos e análise de convergência, Curva de Aprendizagem e Desajuste do referido algoritmo. Dentro do contexto de redes de sensores e filtragem espacial avalia-se o desempenho dos algoritmos por meio da curva de aprendizagem dos referidos algoritmos para os arranjos de antenas adaptativas. No contexto intrínseco da aplicação em engenharia elétrica, isto é, na área de telecomunicações procura-se a melhor alternativa e almeja-se a otimização do processo de transmissão/recepção para eliminar interferências e a menor quantidade de elementos em arranjos de antenas adaptativas, que são conhecidas como antenas inteligentes, e que tem como objetivo atingir uma relação Sinal Ruído para valor pequeno, com número adequado de elementos. O desempenho do algoritmo LMS é avaliado em redes de sensores que é caracterizada por um arranjo de antenas. Resultados de simulações computacionais para diferentes cenários de operação mostram que os algoritmos apresentam bons resultados numéricos de convergência para uma escolha adequada dos parâmetros relacionados com a taxa de aprendizagem que são associadas com suas curvas médias e com a conformação de feixes do arranjo em antenas inteligentes.

Tato práce je psaná v angličtině a je zaměřená na problematiku ekvalizace v digitálních komunikačních systémech. Teoretická část zahrnuje stručné pozorování různých způsobů návrhu ekvalizérů. Praktická část se zabývá implementací nejčastěji používaných ekvalizérů a s jejich adaptačními algoritmy. Cílem praktické části je porovnat jejich charakteristiky a odhalit činitele, které ovlivňují kvalitu ekvalizace. V rámci problematiky ekvalizace jsou prozkoumány tři typy ekvalizérů. Lineární ekvalizér, ekvalizér se zpětnou vazbou a ML (Maximum likelihood) ekvalizér. Každý ekvalizér byl testován na modelu, který simuloval reálnou přenosovou soustavu s komplexním zkreslením, která je složena z útlumu, mezisymbolové interference a aditivního šumu. Na základě implenentace byli určeny charakteristiky ekvalizérů a stanoveno že optimální výkon má ML ekvalizér. Adaptační algoritmy hrají významnou roli ve výkonnosti všech zmíněných ekvalizérů. V práci je nastudována skupina stochastických algoritmů jako algoritmus nejmenších čtverců(LMS), Normalizovaný LMS, Variable step-size LMS a algoritmus RLS jako zástupce deterministického přístupu. Bylo zjištěno, že RLS konverguje mnohem rychleji, než algoritmy založené na LMS. Byly nastudovány činitele, které ovlivnili výkon popisovaných algoritmů. Jedním z důležitých činitelů, který ovlivňuje rychlost konvergence a stabilitu algoritmů LMS je parametr velikosti kroku. Dalším velmi důležitým faktorem je výběr trénovací sekvence. Bylo zjištěno, že velkou nevýhodou algoritmů založených na LMS v porovnání s RLS algoritmy je, že kvalita ekvalizace je velmi závislá na spektrální výkonové hustotě a a trénovací sekvenci.