Active shielding base don implicit and decentralized control

Esta tesis expone un nuevo método para aplicar sistemas de control activo de ruido tradicionales para generar zonas de silencio sin colocar sensores dentro. Primero, se demuestra la factibilidad de usar controladores lineales. Posteriormente, un nuevo concepto es definido, llamado control implícito, que es el caso cuando atenuar la presión acústica en un conjunto de posiciones también implica la reducción de presión en otros lugares. Esta propiedad es usada para controlar la presión en una zona de silencio deseada usando sensores sólo en el entorno de dicha zona. Para este esquema de control el número de sensores y actuadores implican un gran costo computacional. Para reducir las limitaciones computacionales, se propone el control descentralizado y una analogía desde la teoría de juegos, analizando el equilibrio de Nash como el valor de las señales de control después de convergencia.This thesis shows a new method to apply traditional active noise control systems to generate silent zones without locate sensors inside. First, it is demonstrated the feasibility of using linear controllers. Then, it defined a new concept called implicit control, which is the case when attenuating acoustic pressure at a set of locations and it also implies attenuation at other locations. This property is used to control the pressure inside a desired silent zone using sensors only at its boundaries. For this control scheme, the number of sensors and actuators implies high computational cost. In order to reduce hardware limitations, decentralized control and a game theoretical approach are proposed, analyzing the Nash equilibrium as the value of control signals after convergence.Doctor en IngenieríaDoctorad

Doctor of Philosophy

dissertationHearing aids suffer from the problem of acoustic feedback that limits the gain provided by hearing aids. Moreover, the output sound quality of hearing aids may be compromised in the presence of background acoustic noise. Digital hearing aids use advanced signal processing to reduce acoustic feedback and background noise to improve the output sound quality. However, it is known that the output sound quality of digital hearing aids deteriorates as the hearing aid gain is increased. Furthermore, popular subband or transform domain digital signal processing in modern hearing aids introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. In this dissertation, we employ a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. In addition, the method of this dissertation automatically identifies and suppresses residual acoustical feedback components at frequencies that have the potential to drive the system to instability. The suppressed frequency components are monitored and the suppression is removed when such frequencies no longer pose a threat to drive the hearing aid system into instability. Together, the method of this dissertation provides more stable gain over traditional methods by reducing acoustical coupling between the microphone and the loudspeaker of a hearing aid. In addition, the method of this dissertation performs necessary hearing aid signal processing with low-delay characteristics. The central idea for the low-delay hearing aid signal processing is a spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. Finally, the method of this dissertation switches to a least-squares adaptation scheme with linear complexity at the onset of howling. The method adapts to the altered feedback path quickly and allows the patient to not lose perceivable information. The complexity of the least-squares estimate is reduced by reformulating the least-squares estimate into a Toeplitz system and solving it with a direct Toeplitz solver. The increase in stable gain over traditional methods and the output sound quality were evaluated with psychoacoustic experiments on normal-hearing listeners with speech and music signals. The results indicate that the method of this dissertation provides 8 to 12 dB more hearing aid gain than feedback cancelers with traditional fixed gain functions. Furthermore, experimental results obtained with real world hearing aid gain profiles indicate that the method of this dissertation provides less distortion in the output sound quality than classical feedback cancelers, enabling the use of more comfortable style hearing aids for patients with moderate to profound hearing loss. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this dissertation exhibits much smaller forward-path delays with superior howling suppression capability