Análisis espectral de señales cardiacas mediante frecuencia dominante

La Ingeniería Biomédica surgió en la década de 1950 como una fascinante mezcla interdisciplinaria, en la cual la ingeniería, la biología y la medicina aunaban esfuerzos para analizar y comprender distintas enfermedades. Las señales existentes en este área deben ser analizadas e interpretadas, más allá de las capacidades limitadas de la simple vista y la experiencia humana. Aquí es donde el procesamiento digital de la señal se postula como una herramienta indispensable para extraer la información relevante oculta en dichas señales. La electrocardiografía fue una de las primeras áreas en las que se aplicó el procesado digital de señales hace más de 50 años. Las señales electrocardiográficas continúan siendo, a día de hoy, objeto de estudio por parte de cardiólogos e ingenieros. En esta área, las técnicas de procesamiento de señal han ayudado a encontrar información oculta a simple vista que ha cambiado la forma de tratar ciertas enfermedades que fueron ya diagnosticadas previamente. Desde entonces, se han desarrollado numerosas técnicas de procesado de señales electrocardiográficas, pudiéndose resumir estas en tres grandes categorías: análisis tiempo-frecuencia, análisis de organización espacio-temporal y separación de la actividad atrial del ruido y las interferencias. Este proyecto se enmarca dentro de la primera categoría, análisis tiempo-frecuencia, y en concreto dentro de lo que se conoce como análisis de frecuencia dominante, la cual se va a aplicar al análisis de señales de fibrilación auricular. El proyecto incluye una parte teórica de análisis y desarrollo de algoritmos de procesado de señal, y una parte práctica, de programación y simulación con Matlab. Matlab es una de las herramientas fundamentales para el procesamiento digital de señales por ordenador, la cual presenta importantes funciones y utilidades para el desarrollo de proyectos en este campo. Por ello, se ha elegido dicho software como herramienta para la implementación del proyecto. ABSTRACT. Biomedical Engineering emerged in the 1950s as a fascinating interdisciplinary blend, in which engineering, biology and medicine pooled efforts to analyze and understand different diseases. Existing signals in this area should be analyzed and interpreted, beyond the limited capabilities of the naked eye and the human experience. This is where the digital signal processing is postulated as an indispensable tool to extract the relevant information hidden in these signals. Electrocardiography was one of the first areas where digital signal processing was applied over 50 years ago. Electrocardiographic signals remain, even today, the subject of close study by cardiologists and engineers. In this area, signal processing techniques have helped to find hidden information that has changed the way of treating certain diseases that were already previously diagnosed. Since then, numerous techniques have been developed for processing electrocardiographic signals. These methods can be summarized into three categories: time-frequency analysis, analysis of spatio-temporal organization and separation of atrial activity from noise and interferences. This project belongs to the first category, time-frequency analysis, and specifically to what is known as dominant frequency analysis, which is one of the fundamental tools applied in the analysis of atrial fibrillation signals. The project includes a theoretical part, related to the analysis and development of signal processing algorithms, and a practical part, related to programming and simulation using Matlab. Matlab is one of the fundamental tools for digital signal processing, presenting significant functions and advantages for the development of projects in this field. Therefore, we have chosen this software as a tool for project implementation.

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.

Sistema de seguridad basado en la clasificación de eventos sonoros

La teoría de reconocimiento y clasificación de patrones y el aprendizaje automático son actualmente áreas de conocimiento en constante desarrollo y con aplicaciones prácticas en múltiples ámbitos de la industria. El propósito de este Proyecto de Fin de Grado es el estudio de las mismas así como la implementación de un sistema software que dé solución a un problema de clasificación de ruido impulsivo, concretamente mediante el desarrollo de un sistema de seguridad basado en la clasificación de eventos sonoros en tiempo real. La solución será integral, comprendiendo todas las fases del proceso, desde la captación de sonido hasta el etiquetado de los eventos registrados, pasando por el procesado digital de señal y la extracción de características. Para su desarrollo se han diferenciado dos partes fundamentales; una primera que comprende la interfaz de usuario y el procesado de la señal de audio donde se desarrollan las labores de monitorización y detección de ruido impulsivo y otra segunda centrada únicamente en la clasificación de los eventos sonoros detectados, definiendo una arquitectura de doble clasificador donde se determina si los eventos detectados son falsas alarmas o amenazas, etiquetándolos como de un tipo concreto en este segundo caso. Los resultados han sido satisfactorios, mostrando una fiabilidad global en el proceso de entorno al 90% a pesar de algunas limitaciones a la hora de construir la base de datos de archivos de audio, lo que prueba que un dispositivo de seguridad basado en el análisis de ruido ambiente podría incluirse en un sistema integral de alarma doméstico aumentando la protección del hogar. ABSTRACT. Pattern classification and machine learning are currently expertise areas under continuous development and also with extensive applications in many business sectors. The aim of this Final Degree Project is to study them as well as the implementation of software to carry on impulsive noise classification tasks, particularly through the development of a security system based on sound events classification. The solution will go over all process stages, from capturing sound to the labelling of the events recorded, without forgetting digital signal processing and feature extraction, everything in real time. In the development of the Project a distinction has been made between two main parts. The first one comprises the user’s interface and the audio signal processing module, where monitoring and impulsive noise detection tasks take place. The second one is focussed in sound events classification tasks, defining a double classifier architecture where it is determined whether detected events are false alarms or threats, labelling them from a concrete category in the latter case. The obtained results have been satisfactory, with an overall reliability of 90% despite some limitations when building the audio files database. This proves that a safety device based on the analysis of environmental noise could be included in a full alarm system increasing home protection standards.

Compensation of biased excitation effects for MLS-based nonlinear systems' identification

MLS-based identification of nonlinear systems is largely affected by deviations in the excitation signal amenable to the combined effect of DC-offset and an arbitrary gain. These induce orthogonality loss in the MLS filter bank output, thus invalidating the underlying identification construction. In this paper we present a correction algorithm to derive the corrected Volterra kernels from the biased estimations provided by the standard MLS-based procedure.

Quality of experience of adaptive video streaming: investigation in service parameters and subjective quality assessment methodology

The usage of HTTP adaptive streaming (HAS) has become widely spread in multimedia services. Because it allows the service providers to improve the network resource utilization and user׳s Quality of Experience (QoE). Using this technology, the video playback interruption is reduced since the network and server status in addition to capability of user device, all are taken into account by HAS client to adapt the quality to the current condition. Adaptation can be done using different strategies. In order to provide optimal QoE, the perceptual impact of adaptation strategies from point of view of the user should be studied. However, the time-varying video quality due to the adaptation which usually takes place in a long interval introduces a new type of impairment making the subjective evaluation of adaptive streaming system challenging. The contribution of this paper is two-fold: first, it investigates the testing methodology to evaluate HAS QoE by comparing the subjective experimental outcomes obtained from ACR standardized method and a semi-continuous method developed to evaluate the long sequences. In addition, influence of using audiovisual stimuli to evaluate the video-related impairment is inquired. Second, impact of some of the adaptation technical factors including the quality switching amplitude and chunk size in combination with high range of commercial content type is investigated. The results of this study provide a good insight toward achieving appropriate testing method to evaluate HAS QoE, in addition to designing switching strategies with optimal visual quality.

A clustering technique for partial discharge and noise sources identification in power cables by means of waveform parameters

On-line partial discharge (PD) measurements have become a common technique for assessing the insulation condition of installed high voltage (HV) insulated cables. When on-line tests are performed in noisy environments, or when more than one source of pulse-shaped signals are present in a cable system, it is difficult to perform accurate diagnoses. In these cases, an adequate selection of the non-conventional measuring technique and the implementation of effective signal processing tools are essential for a correct evaluation of the insulation degradation. Once a specific noise rejection filter is applied, many signals can be identified as potential PD pulses, therefore, a classification tool to discriminate the PD sources involved is required. This paper proposes an efficient method for the classification of PD signals and pulse-type noise interferences measured in power cables with HFCT sensors. By using a signal feature generation algorithm, representative parameters associated to the waveform of each pulse acquired are calculated so that they can be separated in different clusters. The efficiency of the clustering technique proposed is demonstrated through an example with three different PD sources and several pulse-shaped interferences measured simultaneously in a cable system with a high frequency current transformer (HFCT).

Functional impact of cerebral connections

Cerebral networks are complex sets of connections that resemble a ladder-like web of multiple parallel feedforward, lateral, and feedback connections. This static anatomical description has been pivotal in guiding our understanding of signal processing within cerebral networks. However, measures on both magnitude and functional significance of connections are extremely limited. Here, we compare the anatomically defined strengths of a set of cerebral pathways emerging from the visual middle suprasylvian (MS) cortex of the cat with measures of the functional impact the same region has over distant sites. These functional measures were obtained by analyzing the local and distant effects of MS cooling deactivation on deoxyglucose uptake. Relative to major efferent projections from MS cortex that have a strong influence, projections to early visual processing stages have weaker functional influences than predicted from the anatomy. For higher processing stages, the converse holds: projections from MS cortex have stronger functional influence than predicted from the anatomy. We conclude that these and future functional measures, obtained using the same combination of techniques, will furnish fundamental, new information that complements and extends current models of static cerebral networks, and lead to more realistic models of cerebral network function and component interactions.

Phospholipase C β4 is involved in modulating the visual response in mice

Expression of G protein-regulated phospholipase C (PLC) β4 in the retina, lateral geniculate nucleus, and superior colliculus implies that PLC β4 may play a role in the mammalian visual process. A mouse line that lacks PLC β4 was generated and the physiological significance of PLC β4 in murine visual function was investigated. Behavioral tests using a shuttle box demonstrated that the mice lacking PLC β4 were impaired in their visual processing abilities, whereas they showed no deficit in their auditory abilities. In addition, the PLC β4-null mice showed 4-fold reduction in the maximal amplitude of the rod a- and b-wave components of their electroretinograms relative to their littermate controls. However, recording from single rod photoreceptors did not reveal any significant differences between the PLC β4-null and wild-type littermates, nor were there any apparent differences in retinas examined with light microscopy. While the behavioral and electroretinographic results indicate that PLC β4 plays a significant role in mammalian visual signal processing, isolated rod recording shows little or no apparent deficit, suggesting that the effect of PLC β4 deficiency on the rod signaling pathway occurs at some stage after the initial phototransduction cascade and may require cell–cell interactions between rods and other retinal cells.

The corticofugal system for hearing: Recent progress

Peripheral auditory neurons are tuned to single frequencies of sound. In the central auditory system, excitatory (or facilitatory) and inhibitory neural interactions take place at multiple levels and produce neurons with sharp level-tolerant frequency-tuning curves, neurons tuned to parameters other than frequency, cochleotopic (frequency) maps, which are different from the peripheral cochleotopic map, and computational maps. The mechanisms to create the response properties of these neurons have been considered to be solely caused by divergent and convergent projections of neurons in the ascending auditory system. The recent research on the corticofugal (descending) auditory system, however, indicates that the corticofugal system adjusts and improves auditory signal processing by modulating neural responses and maps. The corticofugal function consists of at least the following subfunctions. (i) Egocentric selection for short-term modulation of auditory signal processing according to auditory experience. Egocentric selection, based on focused positive feedback associated with widespread lateral inhibition, is mediated by the cortical neural net working together with the corticofugal system. (ii) Reorganization for long-term modulation of the processing of behaviorally relevant auditory signals. Reorganization is based on egocentric selection working together with nonauditory systems. (iii) Gain control based on overall excitatory, facilitatory, or inhibitory corticofugal modulation. Egocentric selection can be viewed as selective gain control. (iv) Shaping (or even creation) of response properties of neurons. Filter properties of neurons in the frequency, amplitude, time, and spatial domains can be sharpened by the corticofugal system. Sharpening of tuning is one of the functions of egocentric selection.

Adenylate kinase phosphotransfer communicates cellular energetic signals to ATP-sensitive potassium channels

Transduction of energetic signals into membrane electrical events governs vital cellular functions, ranging from hormone secretion and cytoprotection to appetite control and hair growth. Central to the regulation of such diverse cellular processes are the metabolism sensing ATP-sensitive K+ (KATP) channels. However, the mechanism that communicates metabolic signals and integrates cellular energetics with KATP channel-dependent membrane excitability remains elusive. Here, we identify that the response of KATP channels to metabolic challenge is regulated by adenylate kinase phosphotransfer. Adenylate kinase associates with the KATP channel complex, anchoring cellular phosphotransfer networks and facilitating delivery of mitochondrial signals to the membrane environment. Deletion of the adenylate kinase gene compromised nucleotide exchange at the channel site and impeded communication between mitochondria and KATP channels, rendering cellular metabolic sensing defective. Assigning a signal processing role to adenylate kinase identifies a phosphorelay mechanism essential for efficient coupling of cellular energetics with KATP channels and associated functions.

Scientific bases of human-machine communication by voice.

The scientific bases for human-machine communication by voice are in the fields of psychology, linguistics, acoustics, signal processing, computer science, and integrated circuit technology. The purpose of this paper is to highlight the basic scientific and technological issues in human-machine communication by voice and to point out areas of future research opportunity. The discussion is organized around the following major issues in implementing human-machine voice communication systems: (i) hardware/software implementation of the system, (ii) speech synthesis for voice output, (iii) speech recognition and understanding for voice input, and (iv) usability factors related to how humans interact with machines.

Análise de acetona em ar exalado: desenvolvimento de método eletroanalítico e algoritmo para processamento de sinais

Propõe-se método novo e completo para análise de acetona em ar exalado envolvendo coleta com pré-concentração em água, derivatização química e determinação eletroquímica assistida por novo algoritmo de processamento de sinais. Na literatura recente a acetona expirada vem sendo avaliada como biomarcador para monitoramento não invasivo de quadros clínicos como diabetes e insuficiência cardíaca, daí a importância da proposta. Entre as aminas que reagem com acetona para formar iminas eletroativas, estudadas por polarografia em meados do século passado, a glicina apresentou melhor conjunto de características para a definição do método de determinação por voltametria de onda quadrada sem a necessidade de remoção de oxigênio (25 Hz, amplitude de 20 mV, incremento de 5 mV, eletrodo de gota de mercúrio). O meio reacional, composto de glicina (2 mol·L-1) em meio NaOH (1 mol·L-1), serviu também de eletrólito e o pico de redução da imina em -1,57 V vs. Ag|AgCl constituiu o sinal analítico. Para tratamento dos sinais, foi desenvolvido e avaliado um algoritmo inovador baseado em interpolação de linha base por ajuste de curvas de Bézier e ajuste de gaussiana ao pico. Essa combinação permitiu reconhecimento e quantificação de picos relativamente baixos e largos sobre linha com curvatura acentuada e ruído, situação em que métodos convencionais falham e curvas do tipo spline se mostraram menos apropriadas. A implementação do algoritmo (disponível em http://github.com/batistagl/chemapps) foi realizada utilizando programa open source de álgebra matricial integrado diretamente com software de controle do potenciostato. Para demonstrar a generalidade da extensão dos recursos nativos do equipamento mediante integração com programação externa em linguagem Octave (open source), implementou-se a técnica da cronocoulometria tridimensional, com visualização de resultados já tratados em projeções de malha de perspectiva 3D sob qualquer ângulo. A determinação eletroquímica de acetona em fase aquosa, assistida pelo algoritmo baseado em curvas de Bézier, é rápida e automática, tem limite de detecção de 3,5·10-6 mol·L-1 (0,2 mg·L-1) e faixa linear que atende aos requisitos da análise em ar exalado. O acetaldeído, comumente presente em ar exalado, em especial, após consumo de bebidas alcoólicas, dá origem a pico voltamétrico em -1,40 V, contornando interferência que prejudica vários outros métodos publicados na literatura e abrindo possibilidade de determinação simultânea. Resultados obtidos com amostras reais são concordantes com os obtidos por método espectrofotométrico, em uso rotineiro desde o seu aperfeiçoamento na dissertação de mestrado do autor desta tese. Em relação à dissertação, também se otimizou a geometria do dispositivo de coleta, de modo a concentrar a acetona num volume menor de água gelada e prover maior conforto ao paciente. O método completo apresentado, englobando o dispositivo de amostragem aperfeiçoado e o novo e efetivo algoritmo para tratamento automático de sinais voltamétricos, está pronto para ser aplicado. Evolução para um analisador portátil depende de melhorias no limite de detecção e facilidade de obtenção eletrodos sólidos (impressos) com filme de mercúrio, vez que eletrodos de bismuto ou diamante dopado com boro, entre outros, não apresentaram resposta.

Impact of artifact correction methods on R-R interbeat signals to quantifying heart rate variability (HRV) according to linear and nonlinear methods.

In the analysis of heart rate variability (HRV) are used temporal series that contains the distances between successive heartbeats in order to assess autonomic regulation of the cardiovascular system. These series are obtained from the electrocardiogram (ECG) signal analysis, which can be affected by different types of artifacts leading to incorrect interpretations in the analysis of the HRV signals. Classic approach to deal with these artifacts implies the use of correction methods, some of them based on interpolation, substitution or statistical techniques. However, there are few studies that shows the accuracy and performance of these correction methods on real HRV signals. This study aims to determine the performance of some linear and non-linear correction methods on HRV signals with induced artefacts by quantification of its linear and nonlinear HRV parameters. As part of the methodology, ECG signals of rats measured using the technique of telemetry were used to generate real heart rate variability signals without any error. In these series were simulated missing points (beats) in different quantities in order to emulate a real experimental situation as accurately as possible. In order to compare recovering efficiency, deletion (DEL), linear interpolation (LI), cubic spline interpolation (CI), moving average window (MAW) and nonlinear predictive interpolation (NPI) were used as correction methods for the series with induced artifacts. The accuracy of each correction method was known through the results obtained after the measurement of the mean value of the series (AVNN), standard deviation (SDNN), root mean square error of the differences between successive heartbeats (RMSSD), Lomb\'s periodogram (LSP), Detrended Fluctuation Analysis (DFA), multiscale entropy (MSE) and symbolic dynamics (SD) on each HRV signal with and without artifacts. The results show that, at low levels of missing points the performance of all correction techniques are very similar with very close values for each HRV parameter. However, at higher levels of losses only the NPI method allows to obtain HRV parameters with low error values and low quantity of significant differences in comparison to the values calculated for the same signals without the presence of missing points.

Método de estimativa de torque da articulação do joelho baseada em EMG

Este trabalho apresenta um método de estimativa de torque do joelho baseado em sinais eletromiográficos (EMG) durante terapia de reabilitação robótica. Os EMGs, adquiridos de cinco músculos envolvidos no movimento de flexão e extensão do joelho, são processados para encontrar as ativações musculares. Em seguida, mediante um modelo simples de contração muscular, são calculadas as forças e, usando a geometria da articulação, o torque do joelho. As funções de ativação e contração musculares possuem parâmetros limitados que devem ser calibrados para cada usuário, sendo o ajuste feito mediante a minimização do erro entre o torque estimado e o torque medido na articulação usando a dinâmica inversa. São comparados dois métodos iterativos para funções não-lineares como técnicas de otimização restrita para a calibração dos parâmetros: Gradiente Descendente e Quasi-Newton. O processamento de sinais, calibração de parâmetros e cálculo de torque estimado foram desenvolvidos no software MATLAB®; o cálculo de torque medido foi feito no software OpenSim com sua ferramenta de dinâmica inversa.

Sistema para monitoramento e análise de paisagens acústicas submarinas.

O Monitoramento Acústico Passivo (PAM) submarino refere-se ao uso de sistemas de escuta e gravação subaquática, com o intuito de detectar, monitorar e identificar fontes sonoras através das ondas de pressão que elas produzem. Se diz que é passivo já que tais sistemas unicamente ouvem, sem perturbam o meio ambiente acústico existente, diferentemente de ativos, como os sonares. O PAM submarino tem diversas áreas de aplicação, como em sistemas de vigilância militar, seguridade portuária, monitoramento ambiental, desenvolvimento de índices de densidade populacional de espécies, identificação de espécies, etc. Tecnologia nacional nesta área é praticamente inexistente apesar da sua importância. Neste contexto, o presente trabalho visa contribuir com o desenvolvimento de tecnologia nacional no tema através da concepção, construção e operação de equipamento autônomo de PAM e de métodos de processamento de sinais para detecção automatizada de eventos acústicos submarinos. Foi desenvolvido um equipamento, nomeado OceanPod, que possui características como baixo custo de fabrica¸c~ao, flexibilidade e facilidade de configuração e uso, voltado para a pesquisa científica, industrial e para controle ambiental. Vários protótipos desse equipamento foram construídos e utilizados em missões no mar. Essas jornadas de monitoramento permitiram iniciar a criação de um banco de dados acústico, o qual permitiu fornecer a matéria prima para o teste de detectores de eventos acústicos automatizados e em tempo real. Adicionalmente também é proposto um novo método de detecção-identificação de eventos acústicos, baseado em análise estatística da representação tempo-frequência dos sinais acústicos. Este novo método foi testado na detecção de cetáceos, presentes no banco de dados gerado pelas missões de monitoramento.