Algoritmos genéticos en control activo de ruido

Este proyecto se centra en la implementación de un sistema de control activo de ruido mediante algoritmos genéticos. Para ello, se ha tenido en cuenta el tipo de ruido que se quiere cancelar y el diseño del controlador, parte fundamental del sistema de control. El control activo de ruido sólo es eficaz a bajas frecuencias, hasta los 250 Hz, justo para las cuales los elementos pasivos pierden efectividad, y en zonas o recintos de pequeñas dimensiones y conductos. El controlador ha de ser capaz de seguir todas las posibles variaciones del campo acústico que puedan producirse (variaciones de fase, de frecuencia, de amplitud, de funciones de transferencia electro-acústicas, etc.). Su funcionamiento está basado en algoritmos FIR e IIR adaptativos. La elección de un tipo de filtro u otro depende de características tales como linealidad, causalidad y número de coeficientes. Para que la función de transferencia del controlador siga las variaciones que surgen en el entorno acústico de cancelación, tiene que ir variando el valor de los coeficientes del filtro mediante un algoritmo adaptativo. En este proyecto se emplea como algoritmo adaptativo un algoritmo genético, basado en la selección biológica, es decir, simulando el comportamiento evolutivo de los sistemas biológicos. Las simulaciones se han realizado con dos tipos de señales: ruido de carácter aleatorio (banda ancha) y ruido periódico (banda estrecha). En la parte final del proyecto se muestran los resultados obtenidos y las conclusiones al respecto. Summary. This project is focused on the implementation of an active noise control system using genetic algorithms. For that, it has been taken into account the noise type wanted to be canceled and the controller design, a key part of the control system. The active noise control is only effective at low frequencies, up to 250 Hz, for which the passive elements lose effectiveness, and in small areas or enclosures and ducts. The controller must be able to follow all the possible variations of the acoustic field that might be produced (phase, frequency, amplitude, electro-acoustic transfer functions, etc.). It is based on adaptive FIR and IIR algorithms. The choice of a kind of filter or another depends on characteristics like linearity, causality and number of coefficients. Moreover, the transfer function of the controller has to be changing filter coefficients value thought an adaptive algorithm. In this project a genetic algorithm is used as adaptive algorithm, based on biological selection, simulating the evolutionary behavior of biological systems. The simulations have been implemented with two signal types: random noise (broadband) and periodic noise (narrowband). In the final part of the project the results and conclusions are shown.

Acoustic properties of microperforated panels and their optimizatión by simulated annealing

This thesis investigates the acoustic properties of microperforated panels as an alternative to passive noise control. The first chapters are devoted to the review of analytical models to obtain the acoustic impedance and absorption coefficient of perforated panels. The use of panels perforated with circular holes or with slits is discussed. The theoretical models are presented and some modifications are proposed to improve the modeling of the physical phenomena occurring at the perforations of the panels. The absorption band is widened through the use of multiple layer microperforated panels and/or the combination of a millimetric panel with a porous layer that can be a fibrous material or a nylon mesh. A commercial micrometric mesh downstream a millimetric panel is proposed as a very efficient and low cost solution for controlling noise in reduced spaces. The simulated annealing algorithm is used in order to optimize the panel construction to provide a maximum of absorption in a determined wide band frequency range. Experiments are carried out at normal sound incidence and plane waves. One example is shown for a double layer microperforated panel subjected to grazing flow. A good agreement is achieved between the theory and the experiments. RESUMEN En esta tesis se investigan las propiedades acústicas de paneles micro perforados como una alternativa al control pasivo del ruido. Los primeros capítulos están dedicados a la revisión de los modelos de análisis para obtener la impedancia acústica y el coeficiente de absorción de los paneles perforados. El uso de paneles perforados con agujeros circulares o con ranuras es discutido. Se presentan diferentes modelos y se proponen algunas modificaciones para mejorar la modelización de los fenómenos físicos que ocurren en las perforaciones. La banda de absorción se ensancha a través del uso de capas múltiples de paneles micro perforados y/o la combinación de un panel de perforaciones milimétricas combinado con una capa porosa que puede ser un material fibroso o una malla de nylon. Se propone el uso de una malla micrométrica detrás de un panel milimétrico como una solución económica y eficiente para el control del ruido en espacios reducidos. El algoritmo de recocido simulado se utiliza con el fin de optimizar la construcción de paneles micro perforados para proporcionar un máximo de absorción en una banda determinada frecuencias. Los experimentos se llevan a cabo en la incidencia normal de sonido y ondas planas. Se muestra un ejemplo de panel micro perforado de doble capa sometido a flujo rasante. Se consigue un buen acuerdo entre la teoría y los experimentos.

Influence of loudspeaker directivity on the measurement uncertainty of the acoustic testing of facades.

One of the most significant aspects of a building?s acoustic behavior is the airborne sound insulation of the room façades, since this determines the protection of its inhabitants against environmental noise. For this reason, authorities in most countries have established in their acoustic regulations for buildings the minimum value of sound insulation that must be respected for façades. In order to verify compliance with legal requirements it is usual to perform acoustic measurements in the finished buildings and then compare the measurement results with the established limits. Since there is always a certain measurement uncertainty, this uncertainty must be calculated and taken into account in order to ensure compliance with specifications. The most commonly used method for measuring sound insulation on façades is the so-called Global Loudspeaker Method, specified in ISO 140-5:1998. This method uses a loudspeaker placed outside the building as a sound source. The loudspeaker directivity has a significant influence on the measurement results, and these results may change noticeably by choosing different loudspeakers, even though they all fulfill the directivity requirements of ISO 140-5. This work analyzes the influence of the loudspeaker directivity on the results of façade sound insulation measurement, and determines its contribution to measurement uncertainty. The theoretical analysis is experimentally validated by means of an intermediate precision test according to ISO 5725-3:1994, which compares the values of sound insulation obtained for a façade using various loudspeakers with different directivities. Keywords: Uncertainty, Façade, Insulation

Relationship between objective and subjective indicators in urban soundscape analysis. The case of Cordoba - Argentina

This paper presents the main results of a comparative evaluation of some acoustical parameters with the user´s perception of urban sounds. The study was carried out in three open spaces integrated with different environmental characteristics but similar objective conditions of urban noise. The subjective evaluation was done by means of a survey simultaneously with the objective measurements. The results of the crossed analysis confirmed that in environments with similar noise levels not always exists direct correlation between the objetive indicators and the acoustic comfort of the people. To predict the acustical quality of the soundscape it is necessary to consider aspects such as the background noise and the perception of natural or technological sounds as complements of the general sound level.

Sonophone: Desarrollo y evaluación de un sonómetro profesional para iOS

La medida de la presión sonora es un proceso de extrema importancia para la ingeniería acústica, de aplicación en numerosas áreas de esta disciplina, como la acústica arquitectónica o el control de ruido. Sobre todo en esta última, es necesario poder efectuar medidas precisas en condiciones muy diversas. Por otra parte, la ubicuidad de los dispositivos móviles inteligentes (smartphones, tabletas, etc.), dispositivos que integran potencia de procesado, conectividad, interactividad y una interfaz intuitiva en un tamaño reducido, abre la posibilidad de su uso como sistemas de medida de calidad y de coste bajo. En este Proyecto se pretende utilizar las capacidades de entrada y salida, procesado, conectividad inalámbrica y geolocalización de los dispositivos móviles basados en iOS, en concreto el iPhone, para implementar un sistema de medidas acústicas que iguale o supere las prestaciones de los sonómetros existentes en el mercado. SonoPhone permitirá, mediante la conexión de un micrófono de medida adecuado, la realización de medidas de acuerdo a las normas técnicas en vigor, así como la posibilidad de programar, configurar y almacenar o trasmitir las medidas realizadas, que además estarán geolocalizadas con el GPS integrado en el dispositivo móvil. También se permitirá enviar los datos de la medida a un almacenamiento remoto en la nube. La aplicación tiene una estructura modular en la que un módulo de adquisición de datos lee la señal del micrófono, un back-end efectúa el procesado necesario, y otros módulos permiten la calibración del dispositivo y programar y configurar las medidas, así como su almacenamiento y transmisión en red. Una interfaz de usuario (GUI) permite visualizar las medidas y efectuar las configuraciones deseadas por el usuario, todo ello en tiempo real. Además de implementar la aplicación, se ha realizado una prueba de funcionamiento para determinar si el hardware del iPhone es adecuado para la medida de la presión acústica de acuerdo a las normas internacionales. Sound pressure measurement is an extremely important process in the field of acoustic engineering, with applications in numerous subfields, like for instance building acoustics and noise control, where it is necessary to be able to accurately measure sound pressure in very diverse (and sometimes adverse) conditions. On the other hand, the growing ubiquity of mobile devices such as smartphones or tablets, which combine processing power, connectivity, interactivity and an intuitive interface in a small size, makes it possible to use these devices as quality low-cost measurement systems. This Project aims to use the input-output capabilities of iOS-based mobile devices, in particular the iPhone, together with their processing power, wireless connectivity and geolocation features, to implement an acoustic measurement system that rivals the performance of existing devices. SonoPhone allows, with the addition of an adequate measurement microphone, to carry out measurements that comply with current technical regulations, as well as programming, configuring, storing and transmitting the results of the measurement. These measurements will be geolocated using the integrated GPS, and can be transmitted effortlessly to a remote cloud storage. The application is structured in modular fashion. A data acquisition module reads the signal from the microphone, while a back-end module carries out the necessary processing. Other modules permit the device to be calibrated, or control the configuration of the measurement and its storage or transmission. A Graphical User Interface (GUI) allows visual feedback on the measurement in progress, and provides the user with real-time control over the measurement parameters. Not only an application has been developed; a laboratory test was carried out with the goal of determining if the hardware of the iPhone permits the whole system to comply with international regulations regarding sound level meters.

Cancelación activa del ruido utilizando el KIT TMS320C5515 EZDSP

El control, o cancelación activa de ruido, consiste en la atenuación del ruido presente en un entorno acústico mediante la emisión de una señal igual y en oposición de fase al ruido que se desea atenuar. La suma de ambas señales en el medio acústico produce una cancelación mutua, de forma que el nivel de ruido resultante es mucho menor al inicial. El funcionamiento de estos sistemas se basa en los principios de comportamiento de los fenómenos ondulatorios descubiertos por Augustin-Jean Fresnel, Christiaan Huygens y Thomas Young entre otros. Desde la década de 1930, se han desarrollado prototipos de sistemas de control activo de ruido, aunque estas primeras ideas eran irrealizables en la práctica o requerían de ajustes manuales cada poco tiempo que hacían inviable su uso. En la década de 1970, el investigador estadounidense Bernard Widrow desarrolla la teoría de procesado adaptativo de señales y el algoritmo de mínimos cuadrados LMS. De este modo, es posible implementar filtros digitales cuya respuesta se adapte de forma dinámica a las condiciones variables del entorno. Con la aparición de los procesadores digitales de señal en la década de 1980 y su evolución posterior, se abre la puerta para el desarrollo de sistemas de cancelación activa de ruido basados en procesado de señal digital adaptativo. Hoy en día, existen sistemas de control activo de ruido implementados en automóviles, aviones, auriculares o racks de equipamiento profesional. El control activo de ruido se basa en el algoritmo fxlms, una versión modificada del algoritmo LMS de filtrado adaptativo que permite compensar la respuesta acústica del entorno. De este modo, se puede filtrar una señal de referencia de ruido de forma dinámica para emitir la señal adecuada que produzca la cancelación. Como el espacio de cancelación acústica está limitado a unas dimensiones de la décima parte de la longitud de onda, sólo es viable la reducción de ruido en baja frecuencia. Generalmente se acepta que el límite está en torno a 500 Hz. En frecuencias medias y altas deben emplearse métodos pasivos de acondicionamiento y aislamiento, que ofrecen muy buenos resultados. Este proyecto tiene como objetivo el desarrollo de un sistema de cancelación activa de ruidos de carácter periódico, empleando para ello electrónica de consumo y un kit de desarrollo DSP basado en un procesador de muy bajo coste. Se han desarrollado una serie de módulos de código para el DSP escritos en lenguaje C, que realizan el procesado de señal adecuado a la referencia de ruido. Esta señal procesada, una vez emitida, produce la cancelación acústica. Empleando el código implementado, se han realizado pruebas que generan la señal de ruido que se desea eliminar dentro del propio DSP. Esta señal se emite mediante un altavoz que simula la fuente de ruido a cancelar, y mediante otro altavoz se emite una versión filtrada de la misma empleando el algoritmo fxlms. Se han realizado pruebas con distintas versiones del algoritmo, y se han obtenido atenuaciones de entre 20 y 35 dB medidas en márgenes de frecuencia estrechos alrededor de la frecuencia del generador, y de entre 8 y 15 dB medidas en banda ancha. ABSTRACT. Active noise control consists on attenuating the noise in an acoustic environment by emitting a signal equal but phase opposed to the undesired noise. The sum of both signals results in mutual cancellation, so that the residual noise is much lower than the original. The operation of these systems is based on the behavior principles of wave phenomena discovered by Augustin-Jean Fresnel, Christiaan Huygens and Thomas Young. Since the 1930’s, active noise control system prototypes have been developed, though these first ideas were practically unrealizable or required manual adjustments very often, therefore they were unusable. In the 1970’s, American researcher Bernard Widrow develops the adaptive signal processing theory and the Least Mean Squares algorithm (LMS). Thereby, implementing digital filters whose response adapts dynamically to the variable environment conditions, becomes possible. With the emergence of digital signal processors in the 1980’s and their later evolution, active noise cancellation systems based on adaptive signal processing are attained. Nowadays active noise control systems have been successfully implemented on automobiles, planes, headphones or racks for professional equipment. Active noise control is based on the fxlms algorithm, which is actually a modified version of the LMS adaptive filtering algorithm that allows compensation for the acoustic response of the environment. Therefore it is possible to dynamically filter a noise reference signal to obtain the appropriate cancelling signal. As the noise cancellation space is limited to approximately one tenth of the wavelength, noise attenuation is only viable for low frequencies. It is commonly accepted the limit of 500 Hz. For mid and high frequencies, conditioning and isolating passive techniques must be used, as they produce very good results. The objective of this project is to develop a noise cancellation system for periodic noise, by using consumer electronics and a DSP development kit based on a very-low-cost processor. Several C coded modules have been developed for the DSP, implementing the appropriate signal processing to the noise reference. This processed signal, once emitted, results in noise cancellation. The developed code has been tested by generating the undesired noise signal in the DSP. This signal is emitted through a speaker simulating the noise source to be removed, and another speaker emits an fxlms filtered version of the same signal. Several versions of the algorithm have been tested, obtaining attenuation levels around 20 – 35 dB measured in a tight bandwidth around the generator frequency, or around 8 – 15 dB measured in broadband.

Current Mode with RMS Voltage and Offset Control Loops for a Single-Phase Aircraft Inverter Suitable for Parallel and 3-Phase Operation Modes

Rms voltage regulation may be an attractive possibility for controlling power inverters. Combined with a Hall Effect sensor for current control, it keeps its parallel operation capability while increasing its noise immunity, which may lead to a reduction of the Total Harmonic Distortion (THD). Besides, as voltage regulation is designed in DC, a simple PI regulator can provide accurate voltage tracking. Nevertheless, this approach does not lack drawbacks. Its narrow voltage bandwidth makes transients last longer and it increases the voltage THD when feeding non-linear loads, such as rectifying stages. On the other hand, the implementation can fall into offset voltage error. Furthermore, the information of the output voltage phase is hidden for the control as well, making the synchronization of a 3-phase setup not trivial. This paper explains the concept, design and implementation of the whole control scheme, in an on board inverter able to run in parallel and within a 3-phase setup. Special attention is paid to solve the problems foreseen at implementation level: a third analog loop accounts for the offset level is added and a digital algorithm guarantees 3-phase voltage synchronization.

Contribution of the noise radiated by a single vehicle to the road traffic noise

El ruido del tráfico rodado supone aproximadamente la mitad del ruido global ambiental. Las técnicas de control de ruido habitual en emisión (límites de emisión de vehículos) e inmisión (barreras acústicas, doble acristalamiento) no han sido suficientes para disminuir significativamente las molestias por el tráfico rodado en las últimas tres décadas. El efecto positivo de estas técnicas de control ha sido contrarrestado por el aumento de la densidad del tráfico. Por otra parte, la molestia del ruido del tráfico está altamente correlacionada con los niveles máximos de ruido (MNL), producidos por lo general por conductores agresivos. Sin embargo, los sistemas actuales de medición de ruido de tráfico se basan en una valoración global, por lo que no son capaces de discriminar entre los conductores silenciosos y ruidosos. Por lo tanto, en esta tesis se propone un sistema de medida de ruido en el campo cercano, que es capaz de medir la contribución de cada vehículo individual al ruido del tráfico rodado, permitiendo la detección de los conductores ruidosos. Este trabajo describe también una combinación de investigaciones analíticas y experimentales para la identificación de los conductores responsables de la generación de niveles máximos de ruido. El sistema se basa en dos micrófonos embarcados, uno para el ruido del motor y otro para el ruido de rodadura. Con el fin de relacionar estas mediciones de campo cercano con el ruido de los vehículos radiado al campo lejano, se desarrolla un procedimiento completo para la extrapolación del ruido medido por los micrófonos de campo próximo a las posiciones de campo lejano, usando una combinación de predicción analítica y mediciones experimentales. Las correcciones para los niveles extrapolados se deben a factores atmosféricos, al término de divergencia esférica y a las condiciones de absorción de la superficie de propagación. Para el micrófono situado próximo al motor, es necesario también caracterizar las propiedades acústicas del capó del motor. Ambos niveles de ruido se extrapolan de forma independiente a la posición de campo lejano, donde se realiza una comparación entre la predicción y mediciones para confirmar que la metodología es fiable para estimar el impacto a distancia del ruido de tráfico

Intermittency reinjection probability density function with and without noise effects.

Intermittency phenomenon is a continuous route from regular to chaotic behaviour. Intermittency is an occurrence of a signal that alternates chaotic bursts between quasi-regular periods called laminar phases, driven by the so called reinjection probability density function (RPD). In this paper is introduced a new technique to obtain the RPD for type-II and III intermittency. The new RPD is more general than the classical one and includes the classical RPD as a particular case. The probabilities of the laminar length, the average laminar lengths and the characteristic relations are determined with and without lower bound of the reinjection in agreement with numerical simulations. Finally, it is analyzed the noise effect in intermittency. A method to obtain the noisy RPD is developed extending the procedure used in the noiseless case. The analytical results show a good agreement with numerical simulations.

Bit noise from an optical logic gate with laser diodes

Output bits from an optical logic cell present noise due to the type of technique used to obtain the Boolean functions of two input data bits. We have simulated the behavior of an optically programmable logic cell working with Fabry Perot-laser diodes of the same type employed in optical communications (1550nm) but working here as amplifiers. We will report in this paper a study of the bit noise generated from the optical non-linearity process allowing the Boolean function operation of two optical input data signals. Two types of optical logic cells will be analyzed. Firstly, a classical "on-off" behavior, with transmission operation of LD amplifier and, secondly, a more complicated configuration with two LD amplifiers, one working on transmission and the other one in reflection mode. This last configuration has nonlinear behavior emulating SEED-like properties. In both cases, depending on the value of a "1" input data signals to be processed, a different logic function can be obtained. Also a CW signal, known as control signal, may be apply to fix the type of logic function. The signal to noise ratio will be analyzed for different parameters, as wavelength signals and the hysteresis cycles regions associated to the device, in relation with the signals power level applied. With this study we will try to obtain a better understanding of the possible effects present on an optical logic gate with Laser Diodes.

Optical regeneration based on noise generated in bistable devices: going from 2R to 3R

In this paper we propose to employ an instability that occurs in bistable devices as a control signal at the reception stage to generate the clock signal. One of the adopted configurations is composed of two semiconductor optical amplifiers arranged in a cascaded structure. This configuration has an output equivalent to that obtained from Self-Electrooptic Effect Devices (SEEDs), and it can implement the main Boolean functions of two binary inputs. These outputs, obtained from the addition of two binary signals, show a short spike in the transition from "1" to "2" in the internal processing. A similar result is obtained for a simple semiconductor amplifier with bistable behavior. The paper will show how these structures may help recover clock signals in any optical transmission system

Variable structure control with chattering elimination and guaranteed stability for a generalized T-S model

In this paper, a fuzzy based Variable Structure Control (VSC) with guaranteed stability is presented. The main objective is to obtain an improved performance of highly non-linear unstable systems. The main contribution of this work is that, firstly, new functions for chattering reduction and error convergence without sacrificing invariant properties are proposed, which is considered the main drawback of the VSC control. Secondly, the global stability of the controlled system is guaranteed.The well known weighting parameters approach, is used in this paper to optimize local and global approximation and modeling capability of T-S fuzzy model.A one link robot is chosen as a nonlinear unstable system to evaluate the robustness, effectiveness and remarkable performance of optimization approach and the high accuracy obtained in approximating nonlinear systems in comparison with the original T-S model. Simulation results indicate the potential and generality of the algorithm. The application of the proposed FLC-VSC shows that both alleviation of chattering and robust performance are achieved with the proposed FLC-VSC controller. The effectiveness of the proposed controller is proven in front of disturbances and noise effects.

Characteristic study, its identification and self-tuned approach to control hydro-power plants

The water time constant and mechanical time constant greatly influences the power and speed oscillations of hydro-turbine-generator unit. This paper discusses the turbine power transients in response to different nature and changes in the gate position. The work presented here analyses the characteristics of hydraulic system with an emphasis on changes in the above time constants. The simulation study is based on mathematical first-, second-, third- and fourth-order transfer function models. The study is further extended to identify discrete time-domain models and their characteristic representation without noise and with noise content of 10 & 20 dB signal-to-noise ratio (SNR). The use of self-tuned control approach in minimising the speed deviation under plant parameter changes and disturbances is also discussed.

Metodología para ejecución y control de calidad para ortofotos, mediante combinación de sensor digital con LIDAR en zonas rurales

La utilización de una cámara fotogramétrica digital redunda en el aumento demostrable de calidad radiométrica debido a la mejor relación señal/ruido y a los 12 bits de resolución radiométrica por cada pixel de la imagen. Simultáneamente se consigue un notable ahorro de tiempo y coste gracias a la eliminación de las fases de revelado y escaneado de la película y al aumento de las horas de vuelo por día. De otra parte, el sistema láser aerotransportado (LIDAR - Light Detection and Ranging) es un sistema con un elevado rendimiento y rentabilidad para la captura de datos de elevaciones para generar un modelo digital del terreno (MDT) y también de los objetos sobre el terreno, permitiendo así alcanzar alta precisión y densidad de información. Tanto el sistema LIDAR como el sistema de cámara fotogramétrica digital se combinan con otras técnicas bien conocidas: el sistema de posicionamiento global (GPS - Global Positioning System) y la orientación de la unidad de medida inercial (IMU - Inertial Measure Units), que permiten reducir o eliminar el apoyo de campo y realizar la orientación directa de los sensores utilizando datos de efemérides precisas de los satélites. Combinando estas tecnologías, se va a proponer y poner en práctica una metodología para generación automática de ortofotos en países de América del Sur. Analizando la precisión de dichas ortofotos comparándolas con fuente de mayor exactitud y con las especificaciones técnicas del Plan Nacional de Ortofotografía Aérea (PNOA) se determinará la viabilidad de que dicha metodología se pueda aplicar a zonas rurales. ABSTRACT Using a digital photogrammetric camera results in a demonstrable increase of the radiometric quality due to a better improved signal/noise ratio and the radiometric resolution of 12 bits per pixel of the image. Simultaneously a significant saving of time and money is achieved thanks to the elimination of the developing and film scanning stages, as well as to the increase of flying hours per day. On the other hand, airborne laser system Light Detection and Ranging (LIDAR) is a system with high performance and yield for the acquisition of elevation data in order to generate a digital terrain model (DTM), as well as objects on the ground which allows to achieve high accuracy and data density. Both the LIDAR and the digital photogrammetric camera system are combined with other well known techniques: global positioning system (GPS) and inertial measurement unit (IMU) orientation, which are currently in a mature evolutionary stage, which allow to reduce and/or remove field support and perform a direct guidance of sensors using specific historic data from the satellites. By combining these technologies, a methodology for automatic generation of orthophotos in South American countries will be proposed and implemented. Analyzing the accuracy of these orthophotos comparing them with more accurate sources and technical specifications of the National Aerial Orthophoto (PNOA), the viability of whether this methodology should be applied to rural areas, will be determined.

On-line learning of a fuzzy controller for a precise vehicle cruise control system

Usually, vehicle applications require the use of artificial intelligent techniques to implement control methods, due to noise provided by sensors or the impossibility of full knowledge about dynamics of the vehicle (engine state, wheel pressure or occupiers weight). This work presents a method to on-line evolve a fuzzy controller for commanding vehicles? pedals at low speeds; in this scenario, the slightest alteration in the vehicle or road conditions can vary controller?s behavior in a non predictable way. The proposal adapts singletons positions in real time, and trapezoids used to codify the input variables are modified according with historical data. Experimentation in both simulated and real vehicles are provided to show how fast and precise the method is, even compared with a human driver or using different vehicles.