The objective of the research is to achieve a systematic combination of acoustic echo reduction components that together achieve a robust performance of the MCAEC system as a whole. Conventional approaches to the acoustic echo reduction system typically assume that individual components would perform ideally. For example, the adaptive algorithm for AEC is often developed in the absence of strong near-end signal, the algorithm for RES is often an added module that is developed as a separate noise reduction component, and the decorrelation procedure for MCAEC is yet another add-on module that simply introduces some form of distortion to the reference signal. The main challenge is in designing a consistent criterion across all modules that can be jointly optimized to form a more consistent framework for acoustic echo reduction. The decorrelation procedure can potentially benefit from the system approach as well if it is designed by taking the near-end listener into account. The MCAEC system should be optimized not only for the echo cancellation and suppression performance, but also for the reference signal quality after the added distortion from the decorrelation procedure. Finally, a tuning strategy is presented to jointly optimize the parameters across all modules using object criteria.

Ever since the birth of the telephony system, the problem with echoes, arising from impedance mismatch in 2/4-wire hybrids, or acoustic echoes where a loudspeaker signal is picked up by a closely located microphone, has been ever present. The removal of these echoes is crucial in order to achieve an acceptable audio quality for conversation. Today, the perhaps most common way for echo removal is through cancellation, where an adaptive filter is used to produce an estimated replica of the echo which is then subtracted from the echo-infested signal. Echo cancellation in practice requires extensive control of the filter adaptation process in order to obtain as rapid convergence as possible while also achieving robustness towards disturbances. Moreover, despite the rapid advancement in the computational capabilities of modern digital signal processors there is a constant demand for low-complexity solutions that can be implemented using low power and low cost hardware. This thesis presents low-complexity solutions for echo cancellation related to both the actual filter adaptation process itself as well as for controlling the adaptation process in order to obtain a robust system. Extensive simulations and evaluations using real world recorded signals are used to demonstrate the performance of the proposed solutions.

Há um grande interesse hoje em desenvolver terminais viva-voz que permitam aos participantes de uma conversa à distância contarem com um bom grau de naturalidade e inteligibilidade. O objetivo deste trabalho é apresentar solução para dois impedimentos que surgem quando se deseja projetar um terminal viva-voz para ser utilizado em automóveis: o eco acústico resultante do acoplamento entre microfone e alto-falante do terminal; e o ruído ambiente produzido por exemplo pelo vento, pneus e motor do veículo. A solução proposta envolve o uso de filtros adaptativos e alterações no espectro do sinal de voz para minimizar os problemas mencionados. Os aspectos teóricos são abordados de forma breve, sem deixar no entanto que nenhum detalhe importante fique de fora. Uma implementação prática e eficiente em processador digital de sinais é um dos destaques do trabalho. / There is currently great motivation in developing hands-free devices which offer users, engaged in a telephone conversation, a good level of naturalness and intelligibility. In this work, the goal is to present a solution for two well-known problems that occur when designing a hands-free device for use in automobile environments: (1) the acoustic echo coupling between microphone and speaker, and (2) the background noise generated for example by wind, tires and vehicle engine. The proposed solution includes adaptive filtering techniques and modifications in the speech signal spectrum, in order to minimize the two problems above. Theoretical issues are briefly analyzed, however the author believes no relevant detail is kept out. Highlighted in the report is a practical and efficient implementation of the algorithms in a modern digital signal processor.

Há um grande interesse hoje em desenvolver terminais viva-voz que permitam aos participantes de uma conversa à distância contarem com um bom grau de naturalidade e inteligibilidade. O objetivo deste trabalho é apresentar solução para dois impedimentos que surgem quando se deseja projetar um terminal viva-voz para ser utilizado em automóveis: o eco acústico resultante do acoplamento entre microfone e alto-falante do terminal; e o ruído ambiente produzido por exemplo pelo vento, pneus e motor do veículo. A solução proposta envolve o uso de filtros adaptativos e alterações no espectro do sinal de voz para minimizar os problemas mencionados. Os aspectos teóricos são abordados de forma breve, sem deixar no entanto que nenhum detalhe importante fique de fora. Uma implementação prática e eficiente em processador digital de sinais é um dos destaques do trabalho. / There is currently great motivation in developing hands-free devices which offer users, engaged in a telephone conversation, a good level of naturalness and intelligibility. In this work, the goal is to present a solution for two well-known problems that occur when designing a hands-free device for use in automobile environments: (1) the acoustic echo coupling between microphone and speaker, and (2) the background noise generated for example by wind, tires and vehicle engine. The proposed solution includes adaptive filtering techniques and modifications in the speech signal spectrum, in order to minimize the two problems above. Theoretical issues are briefly analyzed, however the author believes no relevant detail is kept out. Highlighted in the report is a practical and efficient implementation of the algorithms in a modern digital signal processor.

<p>The acoustic echo problem arises whenever there is acoustic coupling between a loudspeaker and a microphone, such as in a teleconference system. This problem is traditionally solved by using an acoustic echo canceler (AEC), which models the echo path with adaptive filters. Long adaptive filters are necessary for satisfactory echo cancellation, which makes AEC highly computationally complex. Recently, a low-complexity echo suppression scheme was presented, the perceptual acoustic echo suppressor (PAES). Spectral modification is used to suppress the echoes, and the complexity is reduced by incorporating perceptual theories. However, under ideal conditions AEC performs better than PAES. </p><p>This thesis considers a hybrid system, which combines AEC and PAES. AEC is used to cancel low-frequency echo components, while PAES suppresses high-frequency echo components. The hybrid system is simulated and assessed, both through subjective listening tests and objective evaluations. The hybrid scheme is shown to have virtually the same perceived quality as a full-band AEC, while having a significantly lower complexity and a higher degree of robustness.</p>

In hands-free telephony and in teleconference systems, the main aim is to provide agood free voice quality when two or more people communicate from different places.The problem often arises during the conversation is the creation of acoustic echo. Thisproblem will cause the bad quality of voice signal and thus talkers could not hearclearly the content of the conversation, even thought lost the important information.This acoustic echo is actually the noise which is created by the reflection of soundwaves by the wall of the room and the other things exist in the room. The mainobjective for engineers is the cancellation of this acoustic echo and provides an echofree environment for speakers during conversation. For this purpose, scientists designdifferent adaptive filter algorithms. Our thesis is also to study and simulate theacoustics echo cancellation by using different adaptive algorithms.

The acoustic echo problem arises whenever there is acoustic coupling between a loudspeaker and a microphone, such as in a teleconference system. This problem is traditionally solved by using an acoustic echo canceler (AEC), which models the echo path with adaptive filters. Long adaptive filters are necessary for satisfactory echo cancellation, which makes AEC highly computationally complex. Recently, a low-complexity echo suppression scheme was presented, the perceptual acoustic echo suppressor (PAES). Spectral modification is used to suppress the echoes, and the complexity is reduced by incorporating perceptual theories. However, under ideal conditions AEC performs better than PAES. This thesis considers a hybrid system, which combines AEC and PAES. AEC is used to cancel low-frequency echo components, while PAES suppresses high-frequency echo components. The hybrid system is simulated and assessed, both through subjective listening tests and objective evaluations. The hybrid scheme is shown to have virtually the same perceived quality as a full-band AEC, while having a significantly lower complexity and a higher degree of robustness.

<p>In hands-free telephony and in teleconference systems, the main aim is to provide agood free voice quality when two or more people communicate from different places.The problem often arises during the conversation is the creation of acoustic echo. Thisproblem will cause the bad quality of voice signal and thus talkers could not hearclearly the content of the conversation, even thought lost the important information.This acoustic echo is actually the noise which is created by the reflection of soundwaves by the wall of the room and the other things exist in the room. The mainobjective for engineers is the cancellation of this acoustic echo and provides an echofree environment for speakers during conversation. For this purpose, scientists designdifferent adaptive filter algorithms. Our thesis is also to study and simulate theacoustics echo cancellation by using different adaptive algorithms.</p>

<p>The complexity of industrial systems and the mathematical models to describe them increases. In many cases, point sensors are no longer sufficient to provide controllers and monitoring instruments with the information necessary for operation. The need for other types of information, such as audio and video, has grown. These are examples of information-rich signals for which suitable applications range in a broad spectrum from micro-electromechanical systems and bio-medical engineering to paper making and steel production.</p><p>Recursive parameter estimation algorithms are employed to identify parameters in a mathematical model from measurements of input and output signals. For accurate parameter estimation, the input signal must be <i>persistently exciting, i.e.</i> such that important features of the modelled system are reflected in the output over a sufficient period of time.</p><p>The Stenlund-Gustafsson (SG) algorithm, a Kalman filter based method for recursive parameter estimation in linear regression models, that does not diverge under lack of excitation, is studied. The stationary properties of the algorithm and the corresponding Riccati equation are formulated in terms of the excitation space spanned by the regressor vectors.</p><p>Furthermore, it is shown that the Riccati equation of the studied algorithm can be solved element-wise. Convergence estimates for the elements of the solution to the Riccati equation are provided, directly relating convergence rate to the signal-to-noise ratio in the regression model. An element-wise form of the parameter update equation is also given, where the connection to specific elements of the solution to the Riccati equation is apparent.</p><p>The SG-algorithm is employed for two applications with audio signals. One is in an acoustic echo cancellation setting where its performance is shown to match that of other well-known estimation techniques, such as the normalized least mean squares and the Kalman filter. When the input is not sufficiently exciting, the studied method performs best of all considered schemes.</p><p>The other application is the Linz-Donawitz (LD) steel converter. The converter consists of a vessel with molten metal and foam is produced to facilitate chemical reactions. A common problem, usually referred to as slopping, arises when the foam rises above the limits of the vessel and overflows. A warning system is designed, based on the SG-algorithm and change detection methods, to give alarms before slopping occurs. A black-box model relates different sensor values of which one is the microphone signal picked up in the area above the converter. The system detected slopping correctly in 80% of the blows in field studies at SSAB Oxelösund.</p><p>A practical example of a vision-based system is provided by cavity form estimation in a water model of the steel bath. The water model is captured from the side by a video camera. The images together with a non-linear model are used to estimate important process parameters, related to the heat and mass transport in the LD-converter.</p>

The complexity of industrial systems and the mathematical models to describe them increases. In many cases, point sensors are no longer sufficient to provide controllers and monitoring instruments with the information necessary for operation. The need for other types of information, such as audio and video, has grown. These are examples of information-rich signals for which suitable applications range in a broad spectrum from micro-electromechanical systems and bio-medical engineering to paper making and steel production. Recursive parameter estimation algorithms are employed to identify parameters in a mathematical model from measurements of input and output signals. For accurate parameter estimation, the input signal must be persistently exciting, i.e. such that important features of the modelled system are reflected in the output over a sufficient period of time. The Stenlund-Gustafsson (SG) algorithm, a Kalman filter based method for recursive parameter estimation in linear regression models, that does not diverge under lack of excitation, is studied. The stationary properties of the algorithm and the corresponding Riccati equation are formulated in terms of the excitation space spanned by the regressor vectors. Furthermore, it is shown that the Riccati equation of the studied algorithm can be solved element-wise. Convergence estimates for the elements of the solution to the Riccati equation are provided, directly relating convergence rate to the signal-to-noise ratio in the regression model. An element-wise form of the parameter update equation is also given, where the connection to specific elements of the solution to the Riccati equation is apparent. The SG-algorithm is employed for two applications with audio signals. One is in an acoustic echo cancellation setting where its performance is shown to match that of other well-known estimation techniques, such as the normalized least mean squares and the Kalman filter. When the input is not sufficiently exciting, the studied method performs best of all considered schemes. The other application is the Linz-Donawitz (LD) steel converter. The converter consists of a vessel with molten metal and foam is produced to facilitate chemical reactions. A common problem, usually referred to as slopping, arises when the foam rises above the limits of the vessel and overflows. A warning system is designed, based on the SG-algorithm and change detection methods, to give alarms before slopping occurs. A black-box model relates different sensor values of which one is the microphone signal picked up in the area above the converter. The system detected slopping correctly in 80% of the blows in field studies at SSAB Oxelösund. A practical example of a vision-based system is provided by cavity form estimation in a water model of the steel bath. The water model is captured from the side by a video camera. The images together with a non-linear model are used to estimate important process parameters, related to the heat and mass transport in the LD-converter.