Integración del descodificador OpenSVC en el DSP del OMAP3530

Hoy en día el uso de dispositivos portátiles multimedia es ya una realidad totalmente habitual. Además, estos dispositivos tienen una capacidad de cálculo y unos recursos gráficos y de memoria altos, tanto es así que por ejemplo en un móvil se pueden reproducir vídeos de muy alta calidad o tener capacidad para manejar entornos 3D. El precio del uso de estos recursos es un mayor consumo de batería que en ocasiones es demasiado alto y acortan en gran medida la vida de la carga útil de la batería. El Grupo de Diseño Electrónico y Microelectrónico de la Universidad Politécnica de Madrid ha abierto una línea de trabajo que busca la optimización del consumo de energía en este tipo de dispositivos, concretamente en el ámbito de la reproducción de vídeo. El enfoque para afrontar la solución del problema se basa en obtener un mayor rendimiento de la batería a costa de disminuir la experiencia multimedia del usuario. De esta manera, cuando la carga de la batería esté por debajo de un determinado umbral mientras el dispositivo esté reproduciendo un vídeo de alta calidad será el dispositivo quien se autoconfigure dinámicamente para consumir menos potencia en esta tarea, reduciendo la tasa de imágenes por segundo o la resolución del vídeo que se descodifica. Además de lo citado anteriormente se propone dividir la descodificación y la representación del vídeo en dos procesadores, uno de propósito general y otro para procesado digital de señal, con esto se consigue que tener la misma capacidad de cálculo que con un solo procesador pero a una frecuencia menor. Para materializar la propuesta se usará la tarjeta BeagleBoard basada en un procesador multinúcleo OMAP3530 de Texas Instrument que contiene dos núcleos: un ARM1 Cortex-A8 y un DSP2 de la familia C6000. Este procesador multinúcleo además permite modificar la frecuencia de reloj y la tensión de alimentación dinámicamente para conseguir reducir de este modo el consumo del terminal. Por otro lado, como reproductor de vídeos se utilizará una versión de MPlayer que integra un descodificador de vídeo escalable que permite elegir dinámicamente la resolución o las imágenes por segundo que se decodifican para posteriormente mostrarlas. Este reproductor se ejecutará en el núcleo ARM pero debido a la alta carga computacional de la descodificación de vídeos, y que el ARM no está optimizado para este tipo de procesado de datos, el reproductor debe encargar la tarea de la descodificación al DSP. El objetivo de este Proyecto Fin de Carrera consiste en que mientras el descodificador de vídeo está ejecutándose en el núcleo DSP y el Mplayer en el núcleo ARM del OMAP3530 se pueda elegir dinámicamente qué parte del vídeo se descodifica, es decir, seleccionar en tiempo real la calidad o capa del vídeo que se quiere mostrar. Haciendo esto, se podrá quitar carga computacional al núcleo ARM y asignársela al DSP el cuál puede procesarla a menor frecuencia para ahorrar batería. 1 ARM: Es una arquitectura de procesadores de propósito general basada en RISC (Reduced Instruction Set Computer). Es desarrollada por la empresa inglesa ARM holdings. 2 DSP: Procesador Digital de Señal (Digital Signal Processor). Es un sistema basado en procesador, el cual está orientado al cálculo matemático a altas velocidad. Generalmente poseen varias unidades aritmético-lógicas (ALUs) para conseguir realizar varias operaciones simultáneamente. SUMMARY. Nowadays, the use of multimedia devices is a well known reality. In addition, these devices have high graphics and calculus performance and a lot of memory as well. In instance, we can play high quality videos and 3D environments in a mobile phone. That kind of use may increase the device's power consumption and make shorter the battery duration. Electronic and Microelectronic Design Group of Technical University of Madrid has a research line which is looking for optimization of power consumption while these devices are playing videos. The solution of this trouble is based on taking more advantage of battery by decreasing multimedia user experience. On this way, when battery charge is under a threshold while device is playing a high quality video the device is going to configure itself dynamically in order to decrease its power consumption by decreasing frame per second rate, video resolution or increasing the noise in the decoded frame. It is proposed splitting decoding and representation tasks in two processors in order to have the same calculus capability with lower frecuency. The first one is specialized in digital signal processing and the other one is a general purpose processor. In order to materialize this proposal we will use a board called BeagleBoard which is based on a multicore processor called OMAP3530 from Texas Instrument. This processor includes two cores: ARM Cortex-A8 and a TMS320C64+ DSP core. Changing clock frequency and supply voltage is allowed by OMAP3530, we can decrease the power consumption on this way. On the other hand, MPlayer will be used as video player. It includes a scalable video decoder which let us changing dynamically the resolution or frames per second rate of the video in order to show it later. This player will be executed by ARM core but this is not optimized for this task, for that reason, DSP core will be used to decoding video. The target of this final career project is being able to choose which part of the video is decoded each moment while decoder is executed by DSP and Mplayer by ARM. It will be able to change in real time the video quality, resolution and frames per second that user want to show. On this way, reducing the computational charge within the processor will be possible.

Caracterización y optimización de un sistema de sonido profesional

El presente proyecto tiene como objeto caracterizar y optimizar un equipo de sonido profesional, entendiendo por “caracterizar” el determinar los atributos particulares de cada uno de los componentes integrados en el sistema, y entendiendo por “optimizar” el hallar la mejor manera de obtener una respuesta plana para todo el rango de frecuencias, libre de distorsión, y en la mayor área posible. El sistema de sonido utilizado pertenece a un grupo musical de directo, por lo que se instala y se configura en cada concierto en función de las características del recinto, sea cerrado o al aire libre. Con independencia de estas particularidades, el sistema completo se divide en dos formaciones, L y R (lado izquierdo y lado derecho del escenario), por lo que cada formación se compone de un procesador digital de la señal, cuatro etapas de amplificación, un sistema line array de ocho unidades, y un conjunto de ocho altavoces de subgraves. Para llevar a cabo el objetivo planteado, se ha dividido el proyecto en las fases que a continuación se describen. En primer lugar, se han realizado, en la cámara anecoica de la EUITT, las medidas que permiten obtener las características de cada uno de los elementos que componen el sistema. Estas medidas se han almacenado en formato ASCII. En segundo lugar, se ha diseñado una interfaz gráfica que permite, utilizando las medidas almacenadas, caracterizar tanto la respuesta individual de cada elemento de la cadena del sistema de sonido como la respuesta combinada de una unidad line array y una unidad de subgraves. La interfaz es interactiva, y tiene además la capacidad de entregar automáticamente los valores de configuración que permiten la optimización del conjunto. Esto es, obtener alineamiento en el rango de frecuencias compartido por ambas unidades. Las medidas realizadas en la cámara anecoica se han utilizado igualmente para modelar el sistema line array al completo y poder realizar simulaciones en campo libre utilizando programas de predicción acústica. Se ha experimentado con los valores de configuración que permiten el alineamiento de los elementos individuales y obtenidos a través de la interfaz desarrollada, para comprobar la validez de los mismos con la formación line array y subgraves al completo. Por otro lado, se han analizado los métodos de optimización de sistemas propuestos por profesionales reconocidos del medio con el objetivo de aplicarlos en un evento real. En la preparación y montaje del evento, se han aplicado los valores de configuración proporcionados por la interfaz, y se ha comprobado la validez de los mismos realizando medidas in situ según los criterios propuestos en los métodos de optimización estudiados. ABSTRACT. This project aims to characterize and optimize a professional sound system. Characterize must be understood as determining the particular attributes of each component integrated in the system; optimize must be understood as finding the best way to get a flat response for all the frequency range, distortion free, in the largest possible area. The sound system under test belongs to a live musical group, so it is setup and configured on each concert depending on the characteristics of the enclosure, whether it’s indoor or outdoor. Apart from these features, the whole system is divided into two clusters, L and R (left and right side of the stage), so that each one is provided with a digital signal processor, four amplification stages, an eight-units line array system, and a set of eight subwoofers . To accomplish the stated objective, the project has been divided into the steps described below. To begin with, measures have been realized in the anechoic chamber of EUITT, which make possible obtaining the characteristics of each of the elements of the system. These measures have been stored in ASCII format. Then, a graphical interface has been designed that allow, using the stored measurements and from graphics, to characterize both the individual response of each element of the string sound system and the combined response of the several elements. The interface is interactive, and also has the ability to automatically deliver the configuration settings that allow the whole optimization. That means to get alignment in the frequency range shared by a line array unit and a subwoofer unit. The measurements made in the anechoic chamber have also been used to model the complete line array system and to perform free-field simulations using acoustical prediction programs. Simulations have been done with the configuration settings that allow the individual elements alignment (provided by the graphical interface developed), in order to check their validity with the full line array and subwoofer systems. On the other hand, analysis about the optimization methods, proposed by renowned professionals of the field, has been made in order to apply them in a real concert. In the setup and assembly of the event, configuration settings provided by the interface have been applied. Their validity has been proved by making measures on-site according to the criteria set in the studied optimization methods.

Diseño e implementación de un controlador de velocidad PID multivariable para un robot con configuración de triciclo

La robótica móvil constituye un área de desarrollo y explotación de interés creciente. Existen ejemplos de robótica móvil de relevancia destacada en el ámbito industrial y se estima un fuerte crecimiento en el terreno de la robótica de servicios. En la arquitectura software de todos los robots móviles suelen aparecer con frecuencia componentes que tienen asignadas competencias de gobierno, navegación, percepción, etcétera, todos ellos de importancia destacada. Sin embargo, existe un elemento, difícilmente prescindible en este tipo de robots, el cual se encarga del control de velocidad del dispositivo en sus desplazamientos. En el presente proyecto se propone desarrollar un controlador PID basado en el modelo y otro no basado en el modelo. Dichos controladores deberán operar en un robot con configuración de triciclo disponible en el Departamento de Sistemas Informáticos y deberán por tanto ser programados en lenguaje C para ejecutar en el procesador digital de señal destinado para esa actividad en el mencionado robot (dsPIC33FJ128MC802). ABSTRACT Mobile robotics constitutes an area of development and exploitation of increasing interest. There are examples of mobile robotics of outstanding importance in industry and strong growth is expected in the field of service robotics. In the software architecture of all mobile robots usually appear components which have assigned competences of government, navigation, perceptionetc., all of them of major importance. However, there is an essential element in this type of robots, which takes care of the speed control. The present project aims to develop a model-based and other non-model-based PID controller. These controllers must operate in a robot with tricycle settings, available from the Department of Computing Systems, and should therefore be programmed in C language to run on the digital signal processor dedicated to that activity in the robot (dsPIC33FJ128MC802).

Development of an open architecture motion control system

A multipurpose open architecture motion control system was developed with three platforms for control and monitoring. The Visual Basic user interface communicated with the operator and gave instructions to the electronic components. The first platform had a BASIC Stamp based controller and three stepping motors. The second platform had a controller, amplifiers and two DC servomotors. The third platform had a DSP module. In this study, each platform was used on machine tools either to move the table or to evaluate the incoming signal. The study indicated that by using advanced microcontrollers, which use high-level languages, motor controllers, DSPs (Digital Signal Processor) and microcomputers, the motion control of different systems could be realized in a short time. Although, the proposed systems had some limitations, their jobs were performed effectively. ^

Efficiency and power density improvement of grid-connected hybrid renewable energy systems utilizing high frequency-based power converters

High efficiency of power converters placed between renewable energy sources and the utility grid is required to maximize the utilization of these sources. Power quality is another aspect that requires large passive elements (inductors, capacitors) to be placed between these sources and the grid. The main objective is to develop higher-level high frequency-based power converter system (HFPCS) that optimizes the use of hybrid renewable power injected into the power grid. The HFPCS provides high efficiency, reduced size of passive components, higher levels of power density realization, lower harmonic distortion, higher reliability, and lower cost. The dynamic modeling for each part in this system is developed, simulated and tested. The steady-state performance of the grid-connected hybrid power system with battery storage is analyzed. Various types of simulations were performed and a number of algorithms were developed and tested to verify the effectiveness of the power conversion topologies. A modified hysteresis-control strategy for the rectifier and the battery charging/discharging system was developed and implemented. A voltage oriented control (VOC) scheme was developed to control the energy injected into the grid. The developed HFPCS was compared experimentally with other currently available power converters. The developed HFPCS was employed inside a microgrid system infrastructure, connecting it to the power grid to verify its power transfer capabilities and grid connectivity. Grid connectivity tests verified these power transfer capabilities of the developed converter in addition to its ability of serving the load in a shared manner. In order to investigate the performance of the developed system, an experimental setup for the HF-based hybrid generation system was constructed. We designed a board containing a digital signal processor chip on which the developed control system was embedded. The board was fabricated and experimentally tested. The system's high precision requirements were verified. Each component of the system was built and tested separately, and then the whole system was connected and tested. The simulation and experimental results confirm the effectiveness of the developed converter system for grid-connected hybrid renewable energy systems as well as for hybrid electric vehicles and other industrial applications.

Development of New Structural Health Monitoring Techniques

During the past two decades, many researchers have developed methods for the detection of structural defects at the early stages to operate the aerospace vehicles safely and to reduce the operating costs. The Surface Response to Excitation (SuRE) method is one of these approaches developed at FIU to reduce the cost and size of the equipment. The SuRE method excites the surface at a series of frequencies and monitors the propagation characteristics of the generated waves. The amplitude of the waves reaching to any point on the surface varies with frequency; however, it remains consistent as long as the integrity and strain distribution on the part is consistent. These spectral characteristics change when cracks develop or the strain distribution changes. The SHM methods may be used for many applications, from the detection of loose screws to the monitoring of manufacturing operations. A scanning laser vibrometer was used in this study to investigate the characteristics of the spectral changes at different points on the parts. The study started with detecting a load on a plate and estimating its location. The modifications on the part with manufacturing operations were detected and the Part-Based Manufacturing Process Performance Monitoring (PbPPM) method was developed. Hardware was prepared to demonstrate the feasibility of the proposed methods in real time. Using low-cost piezoelectric elements and the non-contact scanning laser vibrometer successfully, the data was collected for the SuRE and PbPPM methods. Locational force, loose bolts and material loss could be easily detected by comparing the spectral characteristics of the arriving waves. On-line methods used fast computational methods for estimating the spectrum and detecting the changing operational conditions from sum of the squares of the variations. Neural networks classified the spectrums when the desktop – DSP combination was used. The results demonstrated the feasibility of the SuRE and PbPPM methods.

Development of New Structural Health Monitoring Techniques

During the past two decades, many researchers have developed methods for the detection of structural defects at the early stages to operate the aerospace vehicles safely and to reduce the operating costs. The Surface Response to Excitation (SuRE) method is one of these approaches developed at FIU to reduce the cost and size of the equipment. The SuRE method excites the surface at a series of frequencies and monitors the propagation characteristics of the generated waves. The amplitude of the waves reaching to any point on the surface varies with frequency; however, it remains consistent as long as the integrity and strain distribution on the part is consistent. These spectral characteristics change when cracks develop or the strain distribution changes. The SHM methods may be used for many applications, from the detection of loose screws to the monitoring of manufacturing operations. A scanning laser vibrometer was used in this study to investigate the characteristics of the spectral changes at different points on the parts. The study started with detecting a load on a plate and estimating its location. The modifications on the part with manufacturing operations were detected and the Part-Based Manufacturing Process Performance Monitoring (PbPPM) method was developed. Hardware was prepared to demonstrate the feasibility of the proposed methods in real time. Using low-cost piezoelectric elements and the non-contact scanning laser vibrometer successfully, the data was collected for the SuRE and PbPPM methods. Locational force, loose bolts and material loss could be easily detected by comparing the spectral characteristics of the arriving waves. On-line methods used fast computational methods for estimating the spectrum and detecting the changing operational conditions from sum of the squares of the variations. Neural networks classified the spectrums when the desktop – DSP combination was used. The results demonstrated the feasibility of the SuRE and PbPPM methods.

Time-Reversal Massive Multipath Effect and Bandwidth Heterogeneity

The proliferation of new mobile communication devices, such as smartphones and tablets, has led to an exponential growth in network traffic. The demand for supporting the fast-growing consumer data rates urges the wireless service providers and researchers to seek a new efficient radio access technology, which is the so-called 5G technology, beyond what current 4G LTE can provide. On the other hand, ubiquitous RFID tags, sensors, actuators, mobile phones and etc. cut across many areas of modern-day living, which offers the ability to measure, infer and understand the environmental indicators. The proliferation of these devices creates the term of the Internet of Things (IoT). For the researchers and engineers in the field of wireless communication, the exploration of new effective techniques to support 5G communication and the IoT becomes an urgent task, which not only leads to fruitful research but also enhance the quality of our everyday life. Massive MIMO, which has shown the great potential in improving the achievable rate with a very large number of antennas, has become a popular candidate. However, the requirement of deploying a large number of antennas at the base station may not be feasible in indoor scenarios. Does there exist a good alternative that can achieve similar system performance to massive MIMO for indoor environment? In this dissertation, we address this question by proposing the time-reversal technique as a counterpart of massive MIMO in indoor scenario with the massive multipath effect. It is well known that radio signals will experience many multipaths due to the reflection from various scatters, especially in indoor environments. The traditional TR waveform is able to create a focusing effect at the intended receiver with very low transmitter complexity in a severe multipath channel. TR's focusing effect is in essence a spatial-temporal resonance effect that brings all the multipaths to arrive at a particular location at a specific moment. We show that by using time-reversal signal processing, with a sufficiently large bandwidth, one can harvest the massive multipaths naturally existing in a rich-scattering environment to form a large number of virtual antennas and achieve the desired massive multipath effect with a single antenna. Further, we explore the optimal bandwidth for TR system to achieve maximal spectral efficiency. Through evaluating the spectral efficiency, the optimal bandwidth for TR system is found determined by the system parameters, e.g., the number of users and backoff factor, instead of the waveform types. Moreover, we investigate the tradeoff between complexity and performance through establishing a generalized relationship between the system performance and waveform quantization in a practical communication system. It is shown that a 4-bit quantized waveforms can be used to achieve the similar bit-error-rate compared to the TR system with perfect precision waveforms. Besides 5G technology, Internet of Things (IoT) is another terminology that recently attracts more and more attention from both academia and industry. In the second part of this dissertation, the heterogeneity issue within the IoT is explored. One of the significant heterogeneity considering the massive amount of devices in the IoT is the device heterogeneity, i.e., the heterogeneous bandwidths and associated radio-frequency (RF) components. The traditional middleware techniques result in the fragmentation of the whole network, hampering the objects interoperability and slowing down the development of a unified reference model for the IoT. We propose a novel TR-based heterogeneous system, which can address the bandwidth heterogeneity and maintain the benefit of TR at the same time. The increase of complexity in the proposed system lies in the digital processing at the access point (AP), instead of at the devices' ends, which can be easily handled with more powerful digital signal processor (DSP). Meanwhile, the complexity of the terminal devices stays low and therefore satisfies the low-complexity and scalability requirement of the IoT. Since there is no middleware in the proposed scheme and the additional physical layer complexity concentrates on the AP side, the proposed heterogeneous TR system better satisfies the low-complexity and energy-efficiency requirement for the terminal devices (TDs) compared with the middleware approach.

Implementação de um Módulo de Supervisão para um Sistema de Detecção de Vazamentos em Dutos de Petróleo

The transport of fluids through pipes is used in the oil industry, being the pipelines an important link in the logistics flow of fluids. However, the pipelines suffer deterioration in their walls caused by several factors which may cause loss of fluids to the environment, justifying the investment in techniques and methods of leak detection to minimize fluid loss and environmental damage. This work presents the development of a supervisory module in order to inform to the operator the leakage in the pipeline monitored in the shortest time possible, in order that the operator log procedure that entails the end of the leak. This module is a component of a system designed to detect leaks in oil pipelines using sonic technology, wavelets and neural networks. The plant used in the development and testing of the module presented here was the system of tanks of LAMP, and its LAN, as monitoring network. The proposal consists of, basically, two stages. Initially, assess the performance of the communication infrastructure of the supervisory module. Later, simulate leaks so that the DSP sends information to the supervisory performs the calculation of the location of leaks and indicate to which sensor the leak is closer, and using the system of tanks of LAMP, capture the pressure in the pipeline monitored by piezoresistive sensors, this information being processed by the DSP and sent to the supervisory to be presented to the user in real time

Generic SoC QR Array Processor for Adaptive Beamforming

A generic architecture for implementing a QR array processor in silicon is presented. This improves on previous research by considerably simplifying the derivation of timing schedules for a QR system implemented as a folded linear array, where account has to be taken of processor cell latency and timing at the detailed circuit level. The architecture and scheduling derived have been used to create a generator for the rapid design of System-on-a-Chip (SoC) cores for QR decomposition. This is demonstrated through the design of a single-chip architecture for implementing an adaptive beamformer for radar applications. Published as IEEE Trans Circuits and Systems Part II, Analog and Digital Signal Processing, April 2003 NOT Express Briefs. Parts 1 and II of Journal reorganised since then into Regular Papers and Express briefs

Special issue on design and programming of signal processors for multimedia communication:Special issue on design and programming of signal processors for multimedia communication

The technical challenges in the design and programming of signal processors for multimedia communication are discussed. The development of terminal equipment to meet such demand presents a significant technical challenge, considering that it is highly desirable that the equipment be cost effective, power efficient, versatile, and extensible for future upgrades. The main challenges in the design and programming of signal processors for multimedia communication are, general-purpose signal processor design, application-specific signal processor design, operating systems and programming support and application programming. The size of FFT is programmable so that it can be used for various OFDM-based communication systems, such as digital audio broadcasting (DAB), digital video broadcasting-terrestrial (DVB-T) and digital video broadcasting-handheld (DVB-H). The clustered architecture design and distributed ping-pong register files in the PAC DSP raise new challenges of code generation.

Nonlinear behavioral modeling of mixed signal RF-digital I/O devices

Esta tese investiga a caracterização (e modelação) de dispositivos que realizam o interface entre os domínios digital e analógico, tal como os buffers de saída dos circuitos integrados (CI). Os terminais sem fios da atualidade estão a ser desenvolvidos tendo em vista o conceito de rádio-definido-por-software introduzido por Mitola. Idealmente esta arquitetura tira partido de poderosos processadores e estende a operação dos blocos digitais o mais próximo possível da antena. Neste sentido, não é de estranhar que haja uma crescente preocupação, no seio da comunidade científica, relativamente à caracterização dos blocos que fazem o interface entre os domínios analógico e digital, sendo os conversores digital-analógico e analógico-digital dois bons exemplos destes circuitos. Dentro dos circuitos digitais de alta velocidade, tais como as memórias Flash, um papel semelhante é desempenhado pelos buffers de saída. Estes realizam o interface entre o domínio digital (núcleo lógico) e o domínio analógico (encapsulamento dos CI e parasitas associados às linhas de transmissão), determinando a integridade do sinal transmitido. Por forma a acelerar a análise de integridade do sinal, aquando do projeto de um CI, é fundamental ter modelos que são simultaneamente eficientes (em termos computacionais) e precisos. Tipicamente a extração/validação dos modelos para buffers de saída é feita usando dados obtidos da simulação de um modelo detalhado (ao nível do transístor) ou a partir de resultados experimentais. A última abordagem não envolve problemas de propriedade intelectual; contudo é raramente mencionada na literatura referente à caracterização de buffers de saída. Neste sentido, esta tese de Doutoramento foca-se no desenvolvimento de uma nova configuração de medição para a caracterização e modelação de buffers de saída de alta velocidade, com a natural extensão aos dispositivos amplificadores comutados RF-CMOS. Tendo por base um procedimento experimental bem definido, um modelo estado-da-arte é extraído e validado. A configuração de medição desenvolvida aborda não apenas a integridade dos sinais de saída mas também do barramento de alimentação. Por forma a determinar a sensibilidade das quantias estimadas (tensão e corrente) aos erros presentes nas diversas variáveis associadas ao procedimento experimental, uma análise de incerteza é também apresentada.

Dynamic and isometric protocols of knee extension: Effect of fatigue on the EMG signal

The objective of this study was to analyze the electromyographic (EMG) signal behavior of rectus femoris (RF), vastus medialis (VM), vastus lateralis (VL) and biceps femoris (caput longum) (BFCL) from nine women during fatiguing dynamic and isometric knee extensions tests and to determine their EMGFT (Electromyographic Fatigue Threshold). Surface electrodes, biological signal acquisition module, analogical-digital converter board and specific software were used. The RMS (Root Mean Square) values obtained from concentric phase (80 to 30 degrees) of the dynamic knee extension andfrom isometric contraction were correlated with time on each load by linear regression analysis. The respective slopes were correlated with the correspondent load to determine the EMGFT. Force (Kgf) and median frequency - MF (Hz) obtained during MIVC (Maximal Isometric Voluntary Contraction) performed before and after the fatiguing tests were calculated in Matlab environment. The results demonstrated that the endurance time decreases with higher loads the EMG amplitude increase with time and was greater at higher loads, between muscles in dynamic exercise the RF and VL showed higher slopes, and in isometric exercise the VL showed the same behavior The EMGFT values were similar in both exercises; the force values predominantly decreased after fatiguing tests; however the MF only decreased after some loads. The protocols proposed allowed standardizing protocols at least to induce the fatigue process and to determine the EMGFT as an endurance indicative, which may be used to evaluate the effectiveness of rehabilitative or training interventions indicated to reduce muscle weakness and fatigue.

Discrete wavelet transform signal analyzer

This paper addresses the problem of processing biological data, such as cardiac beats in the audio and ultrasonic range, and on calculating wavelet coefficients in real time, with the processor clock running at a frequency of present application-specified integrated circuits and field programmable gate array. The parallel filter architecture for discrete wavelet transform (DWT) has been improved, calculating the wavelet coefficients in real time with hardware reduced up to 60%. The new architecture, which also processes inverse DWT, is implemented with the Radix-2 or the Booth-Wallace constant multipliers. One integrated circuit signal analyzer in the ultrasonic range, including series memory register banks, is presented. © 2007 IEEE.

Is complex signal processing for bone conduction hearing aids useful?

OBJECTIVES To establish whether complex signal processing is beneficial for users of bone anchored hearing aids. METHODS Review and analysis of two studies from our own group, each comparing a speech processor with basic digital signal processing (either Baha Divino or Baha Intenso) and a processor with complex digital signal processing (either Baha BP100 or Baha BP110 power). The main differences between basic and complex signal processing are the number of audiologist accessible frequency channels and the availability and complexity of the directional multi-microphone noise reduction and loudness compression systems. RESULTS Both studies show a small, statistically non-significant improvement of speech understanding in quiet with the complex digital signal processing. The average improvement for speech in noise is +0.9 dB, if speech and noise are emitted both from the front of the listener. If noise is emitted from the rear and speech from the front of the listener, the advantage of the devices with complex digital signal processing as opposed to those with basic signal processing increases, on average, to +3.2 dB (range +2.3 … +5.1 dB, p ≤ 0.0032). DISCUSSION Complex digital signal processing does indeed improve speech understanding, especially in noise coming from the rear. This finding has been supported by another study, which has been published recently by a different research group. CONCLUSIONS When compared to basic digital signal processing, complex digital signal processing can increase speech understanding of users of bone anchored hearing aids. The benefit is most significant for speech understanding in noise.