Wavelet description of the Glottal Gap

The Glottal Source correlates reconstructed from the phonated parts of voice may render interesting information with applicability in different fields. One of them is defective closure (gap) detection. Through the paper the background to explain the physical foundations of defective gap are reviewed. A possible method to estimate defective gap is also presented based on a Wavelet Description of the Glottal Source. The method is validated using results from the analysis of a gender-balanced speakers database. Normative values for the different parameters estimated are given. A set of study cases with deficient glottal closure is presented and discussed.

Wavelets in High-Capacity DWDM Systems and Modal-Division Multiplexed Transmissions

Optical communications receivers using wavelet signals processing is proposed in this paper for dense wavelength-division multiplexed (DWDM) systems and modal-division multiplexed (MDM) transmissions. The optical signal-to-noise ratio (OSNR) required to demodulate polarization-division multiplexed quadrature phase shift keying (PDM-QPSK) modulation format is alleviated with the wavelet denoising process. This procedure improves the bit error rate (BER) performance and increasing the transmission distance in DWDM systems. Additionally, the wavelet-based design relies on signal decomposition using time-limited basis functions allowing to reduce the computational cost in Digital-Signal-Processing (DSP) module. Attending to MDM systems, a new scheme of encoding data bits based on wavelets is presented to minimize the mode coupling in few-mode (FWF) and multimode fibers (MMF). The Shifted Prolate Wave Spheroidal (SPWS) functions are proposed to reduce the modal interference.

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.

Caracterización de un descodificador HEVC ejecutándose en un DSP

HEVC es el nuevo estándar de codificación de vídeo que está siendo desarrollado conjuntamente por las organizaciones ITU-T Video Coding Experts Group (VCEG) e ISO/IEC Moving Picture Experts Group (MPEG). Su objetivo principal es mejorar la compresión de vídeo, en relación a los actuales estándares. Es común hoy en día, debido a su flexibilidad para aplicaciones de bajo consumo, diseñar sistemas de descodificación de vídeo basados en un procesador digital de señal (DSP). En la mayoría de las veces, los diseños parten de un código creado para ser ejecutado en un ordenador personal y posteriormente se optimizan para tecnología DSP. El objetivo principal de este proyecto es caracterizar el rendimiento de un sistema basado en DSP que ejecute el código de un descodificador de video HEVC. ABSTRACT. HEVC is a new video coding standard which is being developed by both ITU-T Video Coding Experts Group (VCEG) and ISO/IEC Moving Picture Experts Group (MPEG). Its main goal is to improve video compression, compared with the actual standards. It is common practice, because of the flexibility in low power applications, to design video decoding systems using digital signal processors (DSP). Most of the time, these designs start with a code suitable to be executed in personal computers and then it is optimized forDSP technology. The main goal in this final degree project is to characterize the performance of a DSP based system executing an HEVC video decoder.

Integración de RVC-HEVC y OpenHEVC sobre la plataforma MultiDSP C6678

Este proyecto se basa en la integración de funciones optimizadas de OpenHEVC en el códec Reconfigurable Video Coding (RVC) - High Efficiency Video Coding (HEVC). RVC es un framework capaz de generar automáticamente el código que implementa cualquier estándar de video mediante el uso de librerías. Estas librerías contienen la definición de bloques funcionales de los que se componen los distintos estándares de video a implementar. Sin embargo, como desventaja a la facilidad de creación de estándares utilizando este framework, las librerías que utiliza no se encuentran optimizadas. Por ello se pretende que el códec RVC-HEVC sea capaz de realizar llamadas a funciones optimizadas, que para el estudio éstas se encontrarán en la librería OpenHEVC. Por otro lado, estos codificadores de video se pueden encontrar implementados tanto en PCs como en sistemas embebidos. Los Digital Signal Processors (DSPs) son unas plataformas especializadas en el procesamiento digital, teniendo una alta velocidad en el cómputo de operaciones matemáticas. Por ello, para este proyecto se integrará RVC-HEVC con las llamadas a OpenHEVC en una plataforma DSP como la TMS320C6678. Una vez completa la integración se efectuan medidas de eficiencia para ver cómo las llamadas a funciones optimizadas mejoran la velocidad en la decodificación de imágenes. ABSTRACT. This project is based in the integration of optimized functions from OpenHEVC in the RVC-HEVC (Reconfigurable Video Coding- High Efficiency Video Coding) codec. RVC is a framework capable of generating automatically any type of video standard with the use of libraries. Inside these libraries there are the definitions of the functional blocks which make up the different standards, in which for the case of study will be the HEVC standard. Nevertheless, as a downside for the simplicity in producing standards with the RVC tool, these libraries are not optimized. Thus, one of the goals for the project will be to make the RVC-HEVC call optimized functions, in which in this case they will be inside the OpenHEVC library. On the other hand, these video encoders can be implemented both in PCs and embedded systems. The DSPs (Digital Signal Processors) are platforms specialized in digital processing, being able to compute mathematical operations in a short period of time. Consequently, for this project the integration of the RVC-HEVC with calls to the OpenHEVC library will be done in a DSP platform such as a TMS320C6678. Once completed the integration, performance measures will be carried out to evaluate the improvement in the decoding speed obtained when optimized functions are used by the RVC-HEVC.

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.

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.

What does voice-processing technology support today?

This paper describes the state of the art in applications of voice-processing technologies. In the first part, technologies concerning the implementation of speech recognition and synthesis algorithms are described. Hardware technologies such as microprocessors and DSPs (digital signal processors) are discussed. Software development environment, which is a key technology in developing applications software, ranging from DSP software to support software also is described. In the second part, the state of the art of algorithms from the standpoint of applications is discussed. Several issues concerning evaluation of speech recognition/synthesis algorithms are covered, as well as issues concerning the robustness of algorithms in adverse conditions.

Closed-Loop Deep Brain Stimulation: Bidirectional Neuroprosthetics for Tremor and BCI

Thesis (Ph.D.)--University of Washington, 2016-06

Exploiting sensitivity of nonorthogonal joint diagonalisation as a security mechanism in steganography

We have recently proposed the framework of independent blind source separation as an advantageous approach to steganography. Amongst the several characteristics noted was a sensitivity to message reconstruction due to small perturbations in the sources. This characteristic is not common in most other approaches to steganography. In this paper we discuss how this sensitivity relates the joint diagonalisation inside the independent component approach, and reliance on exact knowledge of secret information, and how it can be used as an additional and inherent security mechanism against malicious attack to discovery of the hidden messages. The paper therefore provides an enhanced mechanism that can be used for e-document forensic analysis and can be applied to different dimensionality digital data media. In this paper we use a low dimensional example of biomedical time series as might occur in the electronic patient health record, where protection of the private patient information is paramount.

Experimental comparison of coherent polarization-switched QPSK to polarization-multiplexed QPSK for 10 x 100 km WDM transmission

Polarization-switched quadrature phase-shift keying has been demonstrated experimentally at 40.5Gb/s with a coherent receiver and digital signal processing. Compared to polarization-multiplexed QPSK at the same bit rate, its back-to-back sensitivity at 10-3 bit-error-ratio shows 0.9dB improvement, and it tolerates about 1.6dB higher launch power for 10 × 100km, 50GHz-spaced WDM transmission allowing 1dB penalty in required optical-signal-to-noise ratio relative to back-to-back.

73.7 Tb/s (96 x 3 x 256-Gb/s) mode-division-multiplexed DP-16QAM transmission with inline MM-EDFA

Transmission of a 73.7 Tb/s (96x3x256-Gb/s) DP-16QAM mode-division-multiplexed signal over 119km of few-mode fiber transmission line incorporating an inline multi mode EDFA and a phase plate based mode (de-)multiplexer is demonstrated. Data-aided 6x6 MIMO digital signal processing was used to demodulate the signal. The total demonstrated net capacity, taking into account 20% of FEC-overhead and 7.5% additional overhead (Ethernet and training sequences), is 57.6 Tb/s, corresponding to a spectral efficiency of 12 bits/s/Hz.

Compensation of nonlinear fibre impairments in coherent systems employing spectrally efficient modulation formats

We investigate electronic mitigation of linear and non-linear fibre impairments and compare various digital signal processing techniques, including electronic dispersion compensation (EDC), single-channel back-propagation (SC-BP) and back-propagation with multiple channel processing (MC-BP) in a nine-channel 112 Gb/s PM-mQAM (m=4,16) WDM system, for reaches up to 6,320 km. We show that, for a sufficiently high local dispersion, SC-BP is sufficient to provide a significant performance enhancement when compared to EDC, and is adequate to achieve BER below FEC threshold. For these conditions we report that a sampling rate of two samples per symbol is sufficient for practical SC-BP, without significant penalties.

Channel estimation and compensation in chromatic dispersion limited optical fast OFDM systems

We experimentally investigate the channel estimation and compensation in a chromatic dispersion (CD) limited 20Gbit/s optical fast orthogonal frequency division multiplexing (F-OFDM) system with up to 840km transmission. It is shown that symmetric extension based guard interval (GI) is required to enable CD compensation using one-tap equalizers. As few as one optical F-OFDM symbol with four and six pilot tones per symbol can achieve near-optimal channel estimation and compensation performance for 600km and 840km respectively.

Mode-division-multiplexed 3x112-Gb/s DP-QPSK transmission over 80 km few-mode fiber with inline MM-EDFA and blind DSP

We show transmission of a 3x112-Gb/s DP-QPSK mode-division-multiplexed signal up to 80km, with and without multi-mode EDFA, using blind 6x6 MIMO digital signal processing. We show that the OSNR-penalty induced by mode-mixing in the multi-mode EDFA is negligible.