Vision-based active safety system for automatic stopping

ntelligent systems designed to reduce highway fatalities have been widely applied in the automotive sector in the last decade. Of all users of transport systems, pedestrians are the most vulnerable in crashes as they are unprotected. This paper deals with an autonomous intelligent emergency system designed to avoid collisions with pedestrians. The system consists of a fuzzy controller based on the time-to-collision estimate – obtained via a vision-based system – and the wheel-locking probability – obtained via the vehicle’s CAN bus – that generates a safe braking action. The system has been tested in a real car – a convertible Citroën C3 Pluriel – equipped with an automated electro-hydraulic braking system capable of working in parallel with the vehicle’s original braking circuit. The system is used as a last resort in the case that an unexpected pedestrian is in the lane and all the warnings have failed to produce a response from the driver.

Diseño y test VHDL de un cambiador de formato para ver TV3D en un televisor convencional

La difusión de TV3D actual utiliza formatos como el Side-by-Side o Top-and-Bottom, en los que cada par de imágenes, correspondiente a las vistas de los ojos derecho e izquierdo, se encapsula con la mitad de la resolución espacial en una sola imagen. Estas imágenes se muestran de manera casi simultánea de forma que el ojo humano compone una imagen con profundidad que se asemeja a la visión binocular natural. Desde hace un par de años las principales plataformas de televisión han empezado a crear canales con contenido 3D. La televisión 3D (TV3D) se ha introducido en los hogares gracias a los televisores estereoscópicos. Estos televisores, que son compatibles con los formatos antes mencionados, extraen de cada imagen sus dos vistas, recuperan la resolución original y presentan cada vista alternativamente en la pantalla, generando al mismo tiempo una señal de sincronismo para las gafas activas, creando de esta forma la sensación tridimensional de las imágenes. En este PFC se pretende realizar el diseño VHDL de un cambiador de formato que genere en tiempo real la secuencia de imágenes correspondiente a los ojos derecho e izquierdo, con resolución completa, a partir de una secuencia codificada en formato tipo Top-and-Bottom y el banco de test para su prueba. Este circuito se implementará como un periférico del procesador NIOS II de Altera. El diseño podría utilizarse como parte de un sistema que permita la visualización de las actuales emisiones de televisión 3D en un televisor convencional. La tecnología de referencia que se utilizará serán las FPGAs, más concretamente la tarjeta Cyclone III FPGA Starter Kit (EP3C25 FPGA) de Altera, junto a una tarjeta de ampliación de Microtronix con entrada y salida HDMI para video y audio. Además se pretende crear la documentación necesaria para el desarrollo de futuros trabajos relacionados con la televisión 3D. ABSTRACT Current TV3D broadcasting uses formats as Side-by-Side or Top-and-Bottom, where every single pair of images, corresponding to left and right eyes views, are encapsulated with half spatial resolution in one single image. These images are almost simultaneously displayed so that the human eye forms an image with depth resembling naturally binocular vision. From a couple of years the major TV platforms have begun to create 3D content channels. 3D Television (3DTV) has been introduced in homes through stereoscopic televisions. These televisions, which are compatible with the above formats, each image is extracted from the two views, and recover the original resolution and displays alternately each view in screen, while generating a synchronization signal for active glasses, thereby creating the three-dimensional sensation of the images. The main objective in this PFC is to make the design of an exchanger VHDL format in real time to generate the image sequence corresponding to the right and left eyes, with full resolution from an encoded sequence type format Top-and-Bottom and test bench for testing. This circuit is implemented as a Altera NIOS II processor peripheral.The design could be used as part of a system enabling the display of current television broadcasts 3D on a conventional television. The reference technology that will be use are FPGAs, more specifically Cyclone III FPGA Starter Card Kit (EP3C25 FPGA) Altera, along with an expansion card Microtronix with HDMI input and output video and audio. It also aims to create documentation for the development of future works related to 3D TV.

A model of the mammalian retina and the visual cortex. Disorders of vision

A model of the mammalian retina and the behavior of the first layers in the visual cortex is reported. The building blocks are optically programmable logic cells. A model of the retina, similar to the one reported by Dowling (1987) is presented. From the model of the visual cortex obtained, some types of symmetries and asymmetries are possible to be detected

Correlación entre señales de audio multi-microfónicas

Este proyecto pretende mostrar los desfases existentes entre señales de audio obtenidas de la misma fuente en distintos puntos distanciados entre sí. Para ello nos basamos en el análisis de la correlación de las señales de audio multi-microfónicas, para determinar los retrasos entre dichas señales. Durante las de tres partes diferentes que conforman este proyecto, explicaremos el dónde, cómo y por qué se produce este efecto en este tipo de señales. En la primera se presentan algunos de los conceptos teóricos necesarios para entender el desarrollo posterior, tales como la coherencia y correlación entre señales, los retardos de fase y la importancia del micro-tiempo. Además se explican diversas técnicas microfónicas que se utilizarán en la tercera parte. A lo largo de la segunda, se presenta el software desarrollado para determinar y corregir el retraso entre las señales que se deseen analizar. Para ello se ha escogido la herramienta de programación Matlab, ya que ha sido la más utilizada en la mayoría de las asignaturas que componen la titulación y por ello se posee el suficiente dominio de la misma. Además de presentar el propio software, al final de esta parte hay un manual de usuario del mismo, en el que se explica el manejo para posibles usos futuros por parte de otras personas interesadas. En la última parte se demuestra en varios casos reales, el estudio de la alineación de tomas multi-microfónicas en las cuales se produce en efecto que se intenta detectar y corregir. Aquí se realizan tres estudios de dicho fenómeno. En el primero se emplean señales digitales internas, concretamente ruido blanco, retrasando algunas muestras dichas señales unas de otras, para luego analizarlas con el software desarrollado y comprobar la eficacia del mismo. En el segundo se analizan la señales de audio obtenidas en el estudio de grabación de varios grupos de música moderna, mostrando los resultados del empleo del software en algunas de ellas, tales como las tomas de batería, bajo y guitarra. En el tercero se analizan las señales de audio obtenidas fuera del estudio de grabación, en donde no se dispone de las supuestas condiciones ideales que se tienen en el entorno que rodea a un estudio de grabación (acústicamente hablando). Se utilizan algunas de las técnicas microfónicas explicadas en el último apartado de la parte dedicada a los conceptos teóricos, para la grabación de una orquesta sinfónica, para luego analizar el efecto buscado mediante nuestro software, presentando los resultados obtenidos. De igual manera se realiza en el estudio con una agrupación coral de cuatro voces dentro de una Iglesia. ABSTRACT This project aims to show delays between audio signals obtained from the same source at diferent points spaced apart. To do this we rely on the analysis of the correlation of multi-microphonic audio signals, to determine the delay between these signals. During three diferent parts that make up this project, we will explain where, how and why this effect occurs in this type of signals. At the first part we present some of the theoretical concepts necessary to understand the subsequent development, such as coherence and correlation between signals, phase delays and the importance of micro-time. Also explains several microphone techniques to be used in the third part. During the second, it presents the software developed to determine and correct the delay between the signals that are desired to analyze. For this we have chosen the programming software Matlab , as it has been the most used in the majority of the subjects in the degree and therefore has suficient command of it. Besides presenting the software at the end of this part there is a user manual of it , which explains the handling for future use by other interested people. The last part is shown in several real cases, the study of aligning multi- microphonic sockets in which it is produced in effect trying to detect and correct. This includes three studies of this phenomenon. In the first internal digital signals are used, basically white noise, delaying some samples the signals from each other, then with software developed analyzing and verifying its efectiveness. In the second analyzes the audio signals obtained in the recording studio several contemporary bands, showing the results of using the software in some of them, such as the taking of drums, bass and guitar. In the third analyzes audio signals obtained outside the recording studio, where there are no ideal conditions alleged to have on the environment surrounding a recording studio (acoustically speaking). We use some of the microphone techniques explained in the last paragraph of the section on theoretical concepts, for the recording of a symphony orchestra, and then analyze the effect sought by our software, presenting the results. Similarly, in the study performed with a four-voice choir in a church.

Distributed Web Interface for Real-Time Spatial Audio Reproduction System

SSR es el acrónimo de SoundScape Renderer (tool for real-time spatial audio reproduction providing a variety of rendering algorithms), es un programa escrito en su mayoría en C++. El programa permite al usuario escuchar tanto sonidos grabados con anterioridad como sonidos en directo. El sonido o los sonidos se oirán, desde el punto de vista del oyente, como si el sonido se produjese en el punto que el programa decida, lo interesante de este proyecto es que el sonido podrá cambiar de lugar, moverse, etc. Todo en tiempo real. Esto se consigue sin modificar el sonido al grabarlo pero sí al emitirlo, el programa calcula las variaciones necesarias para que al emitir el sonido al oyente le llegue como si el sonido realmente se generase en un punto del espacio o lo más parecido posible. La sensación de movimiento no deja de ser el punto anterior cambiando de lugar. La idea era crear una aplicación web basada en Canvas de HTML5 que se comunicará con esta interfaz de usuario remota. Así se solucionarían todos los problemas de compatibilidad ya que cualquier dispositivo con posibilidad de visualizar páginas web podría correr una aplicación basada en estándares web, por ejemplo un sistema con Windows o un móvil con navegador. El protocolo debía de ser WebSocket porque es un protocolo HTML5 y ofrece las “garantías” de latencia que una aplicación con necesidades de información en tiempo real requiere. Nos permite una comunicación full-dúplex asíncrona sin mucho payload que es justo lo que se venía a evitar al no usar polling normal de HTML. El problema que surgió fue que la interfaz de usuario de red que tenía el programa no era compatible con WebSocket debido a un handshacking inicial y obligatorio que realiza el protocolo, por lo que se necesitaba otra interfaz de red. Se decidió entonces cambiar a JSON como formato para el intercambio de mensajes. Al final el proyecto comprende no sólo la aplicación web basada en Canvas sino también un servidor funcional y la definición de una nueva interfaz de usuario de red con su protocolo añadido. ABSTRACT. This project aims to become a part of the SSR tool to extend its capabilities in the field of the access. SSR is an acronym for SoundScape Renderer, is a program mostly written in C++ that allows you to hear already recorded or live sound with a variety of sound equipment as if the sound came from a desired place in the space. Like the web-page of the SSR says surely better explained: “The SoundScape Renderer (SSR) is a tool for real-time spatial audio reproduction providing a variety of rendering algorithms.” The application can be used with a graphical interface written in Qt but has also a network interface for external applications to use it. This network interface communicates using XML messages. A good example of it is the Android client. This Android client is already working. In order to use the application should be run it by loading an audio source and the wanted environment so that the renderer knows what to do. In that moment the server binds and anyone can use the network interface. Since the network interface is documented everyone can make an application to interact with this network interface. So the application can have as many user interfaces as wanted. The part that is developed in this project has nothing to do neither with audio rendering nor even with the reproduction of the spatial audio. The part that is developed here is about the interface used in the SSR application. As it can be deduced from the title: “Distributed Web Interface for Real-Time Spatial Audio Reproduction System”, this work aims only to offer the interface via web for the SSR (“Real-Time Spatial Audio Reproduction System”). The idea is not to make a new graphical interface for SSR but to allow more types of interfaces and communication. To accomplish the objective of allowing more graphical interfaces this project is going to use a new network interface. By now the SSR application is using only XML for data interchange but this new network interface support JSON. This project comprehends the server that launch the application, the user interface and the new network interface. It is done with these modules in order to allow creating new user interfaces that can communicate with the server or new servers that can communicate with the user interface by defining a complete network interface for data interchange.

Offshore Measurements of Ocean Waves using Stereo Vision Systems

In recent years, remote sensing imaging systems for the measurement of oceanic sea states have attracted renovated attention. Imaging technology is economical, non-invasive and enables a better understanding of the space-time dynamics of ocean waves over an area rather than at selected point locations of previous monitoring methods (buoys, wave gauges, etc.). We present recent progress in space-time measurement of ocean waves using stereo vision systems on offshore platforms, which focus on sea states with wavelengths in the range of 0.01 m to 10 m. Classical epipolar techniques and modern variational methods are reviewed to reconstruct the sea surface from the stereo pairs sequentially in time. The statistical and spectral properties of the resulting observed waves are analyzed. Current improvements of the variational methods are discussed as future lines of research.

An approach to a model of the visual cortex: disorders of vision

One of the most challenging problems that must be solved by any theoretical model purporting to explain the competence of the human brain for relational tasks is the one related with the analysis and representation of the internal structure in an extended spatial layout of múltiple objects. In this way, some of the problems are related with specific aims as how can we extract and represent spatial relationships among objects, how can we represent the movement of a selected object and so on. The main objective of this paper is the study of some plausible brain structures that can provide answers in these problems. Moreover, in order to achieve a more concrete knowledge, our study will be focused on the response of the retinal layers for optical information processing and how this information can be processed in the first cortex layers. The model to be reported is just a first trial and some major additions are needed to complete the whole vision process.

Offshore remote sensing of the ocean by stereo vision systems

In recent years, remote sensing imaging systems for the measurement of oceanic sea states have attracted renovated attention. Imaging technology is economical, non-invasive and enables a better understanding of the space-time dynamics of ocean waves over an area rather than at selected point locations of previous monitoring methods (buoys, wave gauges, etc.). We present recent progress in space-time measurement of ocean waves using stereo vision systems on offshore platforms, which focus on sea states with wavelengths in the range of 0.01 m to 1 m. Both traditional disparity-based systems and modern elevation-based ones are presented in a variational optimization framework: the main idea is to pose the stereoscopic reconstruction problem of the surface of the ocean in a variational setting and design an energy functional whose minimizer is the desired temporal sequence of wave heights. The functional combines photometric observations as well as spatial and temporal smoothness priors. Disparity methods estimate the disparity between images as an intermediate step toward retrieving the depth of the waves with respect to the cameras, whereas elevation methods estimate the ocean surface displacements directly in 3-D space. Both techniques are used to measure ocean waves from real data collected at offshore platforms in the Black Sea (Crimean Peninsula, Ukraine) and the Northern Adriatic Sea (Venice coast, Italy). Then, the statistical and spectral properties of the resulting observed waves are analyzed. We show the advantages and disadvantages of the presented stereo vision systems and discuss future lines of research to improve their performance in critical issues such as the robustness of the camera calibration in spite of undesired variations of the camera parameters or the processing time that it takes to retrieve ocean wave measurements from the stereo videos, which are very large datasets that need to be processed efficiently to be of practical usage. Multiresolution and short-time approaches would improve efficiency and scalability of the techniques so that wave displacements are obtained in feasible times.

A vision-based strategy for autonomous aerial refueling tasks

Autonomous aerial refueling is a key enabling technology for both manned and unmanned aircraft where extended flight duration or range are required. The results presented within this paper offer one potential vision-based sensing solution, together with a unique test environment. A hierarchical visual tracking algorithm based on direct methods is proposed and developed for the purposes of tracking a drogue during the capture stage of autonomous aerial refueling, and of estimating its 3D position. Intended to be applied in real time to a video stream from a single monocular camera mounted on the receiver aircraft, the algorithm is shown to be highly robust, and capable of tracking large, rapid drogue motions within the frame of reference. The proposed strategy has been tested using a complex robotic testbed and with actual flight hardware consisting of a full size probe and drogue. Results show that the vision tracking algorithm can detect and track the drogue at real-time frame rates of more than thirty frames per second, obtaining a robust position estimation even with strong motions and multiple occlusions of the drogue.

A Hierarchical Tracking Strategy for Vision-Based Applications On-Board UAVs

In this paper, we apply a hierarchical tracking strategy of planar objects (or that can be assumed to be planar) that is based on direct methods for vision-based applications on-board UAVs. The use of this tracking strategy allows to achieve the tasks at real-time frame rates and to overcome problems posed by the challenging conditions of the tasks: e.g. constant vibrations, fast 3D changes, or limited capacity on-board. The vast majority of approaches make use of feature-based methods to track objects. Nonetheless, in this paper we show that although some of these feature-based solutions are faster, direct methods can be more robust under fast 3D motions (fast changes in position), some changes in appearance, constant vibrations (without requiring any specific hardware or software for video stabilization), and situations in which part of the object to track is outside of the field of view of the camera. The performance of the proposed tracking strategy on-board UAVs is evaluated with images from realflight tests using manually-generated ground truth information, accurate position estimation using a Vicon system, and also with simulated data from a simulation environment. Results show that the hierarchical tracking strategy performs better than wellknown feature-based algorithms and well-known configurations of direct methods, and that its performance is robust enough for vision-in-the-loop tasks, e.g. for vision-based landing tasks.

Vibration reduction for vision systems on board unmanned aerial vehicles using a neuro-fuzzy controller

In this paper, an intelligent control approach based on neuro-fuzzy systems performance is presented, with the objective of counteracting the vibrations that affect the low-cost vision platform onboard an unmanned aerial system of rotating nature. A scaled dynamical model of a helicopter is used to simulate vibrations on its fuselage. The impact of these vibrations on the low-cost vision system will be assessed and an intelligent control approach will be derived in order to reduce its detrimental influence. Different trials that consider a neuro-fuzzy approach as a fundamental part of an intelligent semi-active control strategy have been carried out. Satisfactory results have been achieved compared to those obtained by means of vibration reduction passive techniques.

OMNIWORKS: Omnidirectional vision for human-UAV co-working

The OMNIWORKS project objective is to develop an autonomous and modular aerial inspection system for an off-shore meteorological mast up to 90m in length. The UAV was equipped with an omni-directional camera and vertical take-off/landing capabilities that should be simple enough to operate as to not need the interventions of a professional pilot under challenging situations. Therefore the tests included different aspects used to evaluate both the technical performance of the UAV behavior as well as the operators? point of view.

Space-time ocean wave measurement using variational stereo vision systems

Remote sensing imaging systems for the measurement of oceanic sea states have recently attracted renovated attention. Imaging technology is economical, non-invasive and enables a better understanding of the space-time dynamics of ocean waves over an area rather than at selected point locations of previous monitoring methods (buoys, wave gauges, etc.). We present recent progress in space-time measurement of ocean waves using stereo vision systems on offshore platforms. Both traditional disparity-based systems and modern elevation-based ones are presented in a variational optimization framework: the main idea is to pose the stereoscopic reconstruction problem of the surface of the ocean in a variational setting and design an energy functional whose minimizer is the desired temporal sequence of wave heights. The functional combines photometric observations as well as spatial and temporal smoothness priors. Disparity methods estimate the disparity between images as an intermediate step toward retrieving the depth of the waves with respect to the cameras, whereas elevation methods estimate the ocean surface displacements directly in 3-D space. Both techniques are used to measure ocean waves from real data collected at offshore platforms in the Black Sea (Crimean Peninsula, Ukraine) and the Northern Adriatic Sea (Venice coast, Italy). Then, the statistical and spectral properties of the resulting observed waves are analyzed. We show the advantages and disadvantages of the presented stereo vision systems and discuss the improvement of their performance in critical issues such as the robustness of the camera calibration in spite of undesired variations of the camera parameters.

Synchronizing sound from different devices over a TCP network

Nowadays, we can send audio on the Internet for multiples uses like telephony, broadcast audio or teleconferencing. The issue comes when you need to synchronize the sound from different sources because the network where we are going to work could lose packets and introduce delay in the delivery. This can also come because the sound cards could be work in different speeds. In this project, we will work with two computers emitting sound (one will simulate the left channel (mono) of a stereo signal, and the other the right channel) and connected with a third computer by a TCP network. The last computer must get the sound from both computers and reproduce it in a speaker properly (without delay). So, basically, the main goal of the project is to synchronize multi-track sound over a network. TCP networks introduce latency into data transfers. Streaming audio suffers from two problems: a delay and an offset between the channels. This project explores the causes of latency, investigates the affect of the inter-channel offset and proposes a solution to synchronize the received channels. In conclusion, a good synchronization of the sound is required in a time when several audio applications are being developed. When two devices are ready to send audio over a network, this multi-track sound will arrive at the third computer with an offset giving a negative effect to the listener. This project has dealt with this offset achieving a good synchronization of the multitrack sound getting a good effect on the listener. This was achieved thanks to the division of the project into several steps having constantly a good vision of the problem, a good scalability and having controlled the latency at all times. As we can see in the chapter 4 of the project, a lack of synchronization over c. 100μs is audible to the listener. RESUMEN. A día de hoy, podemos transmitir audio a través de Internet por varios motivos como pueden ser: una llamada telefónica, una emisión de audio o una teleconferencia. El problema viene cuando necesitas sincronizar ese sonido producido por los diferentes orígenes ya que la red a la que nos vamos a conectar puede perder los paquetes y/o introducir un retardo en las entregas de los mismos. Así mismo, estos retardos también pueden venir producidos por las diferentes velocidades a las que trabajan las tarjetas de sonido de cada dispositivo. En este proyecto, se ha trabajado con dos ordenadores emitiendo sonido de manera intermitente (uno se encargará de simular el canal izquierdo (mono) de la señal estéreo emitida, y el otro del canal derecho), estando conectados a través de una red TCP a un tercer ordenador, el cual debe recibir el sonido y reproducirlo en unos altavoces adecuadamente y sin retardo (deberá juntar los dos canales y reproducirlo como si de estéreo de tratara). Así, el objetivo principal de este proyecto es el de encontrar la manera de sincronizar el sonido producido por los dos ordenadores y escuchar el conjunto en unos altavoces finales. Las redes TCP introducen latencia en la transferencia de datos. El streaming de audio emitido a través de una red de este tipo puede sufrir dos grandes contratiempos: retardo y offset, los dos existentes en las comunicaciones entre ambos canales. Este proyecto se centra en las causas de ese retardo, investiga el efecto que provoca el offset entre ambos canales y propone una solución para sincronizar los canales en el dispositivo receptor. Para terminar, una buena sincronización del sonido es requerida en una época donde las aplicaciones de audio se están desarrollando continuamente. Cuando los dos dispositivos estén preparados para enviar audio a través de la red, la señal de sonido multi-canal llegará al tercer ordenador con un offset añadido, por lo que resultará en una mala experiencia en la escucha final. En este proyecto se ha tenido que lidiar con ese offset mencionado anteriormente y se ha conseguido una buena sincronización del sonido multi-canal obteniendo un buen efecto en la escucha final. Esto ha sido posible gracias a una división del proyecto en diversas etapas que proporcionaban la facilidad de poder solucionar los errores en cada paso dando una importante visión del problema y teniendo controlada la latencia en todo momento. Como se puede ver en el capítulo 4 del proyecto, la falta de sincronización sobre una diferencia de 100μs entre dos canales (offset) empieza a ser audible en la escucha final.

Unidad de efectos de audio en Simulink

La tecnología moderna de computación ha permitido cambiar radicalmente la investigación tecnológica en todos los ámbitos. El proceso general utilizado previamente consistía en el desarrollo de prototipos analógicos, creando múltiples versiones del mismo hasta llegar al resultado adecuado. Este es un proceso costoso a nivel económico y de carga de trabajo. Es por ello por lo que el proceso de investigación actual aprovecha las nuevas tecnologías para lograr el objetivo final mediante la simulación. Gracias al desarrollo de software para la simulación de distintas áreas se ha incrementado el ritmo de crecimiento de los avances tecnológicos y reducido el coste de los proyectos en investigación y desarrollo. La simulación, por tanto, permite desarrollar previamente prototipos simulados con un coste mucho menor para así lograr un producto final, el cual será llevado a cabo en su ámbito correspondiente. Este proceso no sólo se aplica en el caso de productos con circuitería, si bien es utilizado también en productos programados. Muchos de los programas actuales trabajan con algoritmos concretos cuyo funcionamiento debe ser comprobado previamente, para después centrarse en la codificación del mismo. Es en este punto donde se encuentra el objetivo de este proyecto, simular algoritmos de procesado digital de la señal antes de la codificación del programa final. Los sistemas de audio están basados en su totalidad en algoritmos de procesado de la señal, tanto analógicos como digitales, siendo estos últimos los que están sustituyendo al mundo analógico mediante los procesadores y los ordenadores. Estos algoritmos son la parte más compleja del sistema, y es la creación de nuevos algoritmos la base para lograr sistemas de audio novedosos y funcionales. Se debe destacar que los grupos de desarrollo de sistemas de audio presentan un amplio número de miembros con cometidos diferentes, separando las funciones de programadores e ingenieros de la señal de audio. Es por ello por lo que la simulación de estos algoritmos es fundamental a la hora de desarrollar nuevos y más potentes sistemas de audio. Matlab es una de las herramientas fundamentales para la simulación por ordenador, la cual presenta utilidades para desarrollar proyectos en distintos ámbitos. Sin embargo, en creciente uso actualmente se encuentra el software Simulink, herramienta especializada en la simulación de alto nivel que simplifica la dificultad de la programación en Matlab y permite desarrollar modelos de forma más rápida. Simulink presenta una completa funcionalidad para el desarrollo de algoritmos de procesado digital de audio. Por ello, el objetivo de este proyecto es el estudio de las capacidades de Simulink para generar sistemas de audio funcionales. A su vez, este proyecto pretende profundizar en los métodos de procesado digital de la señal de audio, logrando al final un paquete de sistemas de audio compatible con los programas de edición de audio actuales. ABSTRACT. Modern computer technology has dramatically changed the technological research in multiple areas. The overall process previously used consisted of the development of analog prototypes, creating multiple versions to reach the proper result. This is an expensive process in terms of an economically level and workload. For this reason actual investigation process take advantage of the new technologies to achieve the final objective through simulation. Thanks to the software development for simulation in different areas the growth rate of technological progress has been increased and the cost of research and development projects has been decreased. Hence, simulation allows previously the development of simulated protoypes with a much lower cost to obtain a final product, which will be held in its respective field. This process is not only applied in the case of circuitry products, but is also used in programmed products. Many current programs work with specific algorithms whose performance should be tested beforehand, which allows focusing on the codification of the program. This is the main point of this project, to simulate digital signal processing algorithms before the codification of the final program. Audio systems are entirely based on signal processing, both analog and digital systems, being the digital systems which are replacing the analog world thanks to the processors and computers. This algorithms are the most complex part of every system, and the creation of new algorithms is the most important step to achieve innovative and functional new audio systems. It should be noted that development groups of audio systems have a large number of members with different roles, separating them into programmers and audio signal engineers. For this reason, the simulation of this algorithms is essential when developing new and more powerful audio systems. Matlab is one of the most important tools for computer simulation, which has utilities to develop projects in different areas. However, the use of the Simulink software is constantly growing. It is a simulation tool specialized in high-level simulations which simplifies the difficulty of programming in Matlab and allows the developing of models faster. Simulink presents a full functionality for the development of algorithms for digital audio processing. Therefore, the objective of this project is to study the posibilities of Simulink to generate funcional audio systems. In turn, this projects aims to get deeper into the methods of digital audio signal processing, making at the end a software package of audio systems compatible with the current audio editing software.