Diseño y banco de test VHDL de un procesador que realice la ICT para HEVC

Se va a realizar un estudio de la codificación de imágenes sobre el estándar HEVC (high-effiency video coding). El proyecto se va a centrar en el codificador híbrido, más concretamente sobre la aplicación de la transformada inversa del coseno que se realiza tanto en codificador como en el descodificador. La necesidad de codificar vídeo surge por la aparición de la secuencia de imágenes como señales digitales. El problema principal que tiene el vídeo es la cantidad de bits que aparecen al realizar la codificación. Como consecuencia del aumento de la calidad de las imágenes, se produce un crecimiento exponencial de la cantidad de información a codificar. La utilización de las transformadas al procesamiento digital de imágenes ha aumentado a lo largo de los años. La transformada inversa del coseno se ha convertido en el método más utilizado en el campo de la codificación de imágenes y video. Las ventajas de la transformada inversa del coseno permiten obtener altos índices de compresión a muy bajo coste. La teoría de las transformadas ha mejorado el procesamiento de imágenes. En la codificación por transformada, una imagen se divide en bloques y se identifica cada imagen a un conjunto de coeficientes. Esta codificación se aprovecha de las dependencias estadísticas de las imágenes para reducir la cantidad de datos. El proyecto realiza un estudio de la evolución a lo largo de los años de los distintos estándares de codificación de video. Se analiza el codificador híbrido con más profundidad así como el estándar HEVC. El objetivo final que busca este proyecto fin de carrera es la realización del núcleo de un procesador específico para la ejecución de la transformada inversa del coseno en un descodificador de vídeo compatible con el estándar HEVC. Es objetivo se logra siguiendo una serie de etapas, en las que se va añadiendo requisitos. Este sistema permite al diseñador hardware ir adquiriendo una experiencia y un conocimiento más profundo de la arquitectura final. ABSTRACT. A study about the codification of images based on the standard HEVC (high-efficiency video coding) will be developed. The project will be based on the hybrid encoder, in particular, on the application of the inverse cosine transform, which is used for the encoder as well as for the decoder. The necessity of encoding video arises because of the appearance of the sequence of images as digital signals. The main problem that video faces is the amount of bits that appear when making the codification. As a consequence of the increase of the quality of the images, an exponential growth on the quantity of information that should be encoded happens. The usage of transforms to the digital processing of images has increased along the years. The inverse cosine transform has become the most used method in the field of codification of images and video. The advantages of the inverse cosine transform allow to obtain high levels of comprehension at a very low price. The theory of the transforms has improved the processing of images. In the codification by transform, an image is divided in blocks and each image is identified to a set of coefficients. This codification takes advantage of the statistic dependence of the images to reduce the amount of data. The project develops a study of the evolution along the years of the different standards in video codification. In addition, the hybrid encoder and the standard HEVC are analyzed more in depth. The final objective of this end of degree project is the realization of the nucleus from a specific processor for the execution of the inverse cosine transform in a decoder of video that is compatible with the standard HEVC. This objective is reached following a series of stages, in which requirements are added. This system allows the hardware designer to acquire a deeper experience and knowledge of the final architecture.

JChord : An Android application helping people learn guitar chords

Nowadays, a lot of interesting and useful and imaginative applications are springing to Android software market. And for guitar fans, some related apps bring great connivence to them, like a guitar tuner can save people from carrying a entity tuner all the time, some apps can simulate a real guitar, and some apps provide some simple lessons allowing people to learn some basic things. But these apps which can teach people, they can't really “monitor ” people, that is, they just give some instructions and hope people would follow them. So my project is to design an app which can detect if users are playing wrong and right real-timely. Guitar chords are always the first for new guitar beginners to learn, and a chord is a set of notes combined together in a regulated way ( get from the music theory having millions of developing ), and 'pitch' is the term for determining if the note different from other notes or noise, so the problem here is to manage the multi-pitch analysis in real time. And it's necessary to know some basics of digital signal processing ( DSP ) because digital signals are always more convenient for computers to analyze compared to analog signals. Then I found an audio processing Java library – TarsosDSP, and try to apply it to my Android project.

Noise reduction in speech signals using a TMS320C31 digital signal processor

This paper describes a speech enhancement system (SES) based on a TMS320C31 digital signal processor (DSP) for real-time application. The SES algorithm is based on a modified spectral subtraction method and a new speech activity detector (SAD) is used. The system presents a medium computational load and a sampling rate up to 18 kHz can be used. The goal is load and a sampling rate up to 18 kHz can be used. The goal is to use it to reduce noise in an analog telephone line.

Secure transmission of information by digital chaotic signals

Protecting signals is one of the main tasks in information transmission. A large number of different methods have been employed since many centuries ago. Most of them have been based on the use of certain signal added to the original one. When the composed signal is received, if the added signal is known, the initial information may be obtained. The main problem is the type of masking signal employed. One possibility is the use of chaotic signals, but they have a first strong limitation: the need to synchronize emitter and receiver. Optical communications systems, based on chaotic signals, have been proposed in a large number of papers. Moreover, because most of the communication systems are digital and conventional chaos generators are analogue, a conversion analogue-digital is needed. In this paper we will report a new system where the digital chaos is obtained from an optically programmable logic structure. This structure has been employed by the authors in optical computing and some previous results in chaotic signals have been reported. The main advantage of this new system is that an analogue-digital conversion is not needed. Previous works by the authors employed Self-Electrooptical Effect Devices but in this case more conventional structures, as semiconductor laser amplifiers, have been employed. The way to analyze the characteristics of digital chaotic signals will be reported as well as the method to synchronize the chaos generators located in the emitter and in the receiver.

Analysis of digital chaotic optical signals

The main objective of this paper is to present some tools to analyze a digital chaotic signal. We have proposed some of them previously, as a new type of phase diagrams with binary signals converted to hexadecimal. Moreover, the main emphasis will be given in this paper to an analysis of the chaotic signal based on the Lempel and Ziv method. This technique has been employed partly by us to a very short stream of data. In this paper we will extend this method to long trains of data (larger than 2000 bit units). The main characteristics of the chaotic signal are obtained with this method being possible to present numerical values to indicate the properties of the chaos.

Transmission of digital chaotic and information-bearing signals in optical communication systems

A new proposal to have secure communications in a system is reported. The basis is the use of a synchronized digital chaotic systems, sending the information signal added to an initial chaos. The received signal is analyzed by another chaos generator located at the receiver and, by a logic boolean function of the chaotic and the received signals, the original information is recovered. One of the most important facts of this system is that the bandwidth needed by the system remain the same with and without chaos.

INDIRAD : a digital receiver for amplitude modulated signals using Indicator processing /

Thesis (M.S.)--University of Illinois at Urbana-Champaign.

Digital frequency relaying based on the modified least mean square method

This research presents a method for frequency estimation in power systems using an adaptive filter based on the Least Mean Square Algorithm (LMS). In order to analyze a power system, three-phase voltages were converted into a complex signal applying the alpha beta-transform and the results were used in an adaptive filtering algorithm. Although the use of the complex LMS algorithm is described in the literature, this paper deals with some practical aspects of the algorithm implementation. In order to reduce computing time, a coefficient generator was implemented. For the algorithm validation, a computing simulation of a power system was carried Out using the ATP software. Many different situations were Simulated for the performance analysis of the proposed methodology. The results were compared to a commercial relay for validation, showing the advantages of the new method. (C) 2009 Elsevier Ltd. All rights reserved.

A uniform white-noise model for fixed-point roundoff errors in digital systems

Fixed-point roundoff noise in digital implementation of linear systems arises due to overflow, quantization of coefficients and input signals, and arithmetical errors. In uniform white-noise models, the last two types of roundoff errors are regarded as uniformly distributed independent random vectors on cubes of suitable size. For input signal quantization errors, the heuristic model is justified by a quantization theorem, which cannot be directly applied to arithmetical errors due to the complicated input-dependence of errors. The complete uniform white-noise model is shown to be valid in the sense of weak convergence of probabilistic measures as the lattice step tends to zero if the matrices of realization of the system in the state space satisfy certain nonresonance conditions and the finite-dimensional distributions of the input signal are absolutely continuous.

Projecto e implementação de osciloscópio digital baseado na norma IEEE1451.0

Os osciloscópios digitais são utilizados em diversas áreas do conhecimento, assumindo-se no âmbito da engenharia electrónica, como instrumentos indispensáveis. Graças ao advento das Field Programmable Gate Arrays (FPGAs), os instrumentos de medição reconfiguráveis, dadas as suas vantagens, i.e., altos desempenhos, baixos custos e elevada flexibilidade, são cada vez mais uma alternativa aos instrumentos tradicionalmente usados nos laboratórios. Tendo como objectivo a normalização no acesso e no controlo deste tipo de instrumentos, esta tese descreve o projecto e implementação de um osciloscópio digital reconfigurável baseado na norma IEEE 1451.0. Definido de acordo com uma arquitectura baseada nesta norma, as características do osciloscópio são descritas numa estrutura de dados denominada Transducer Electronic Data Sheet (TEDS), e o seu controlo é efectuado utilizando um conjunto de comandos normalizados. O osciloscópio implementa um conjunto de características e funcionalidades básicas, todas verificadas experimentalmente. Destas, destaca-se uma largura de banda de 575kHz, um intervalo de medição de 0.4V a 2.9V, a possibilidade de se definir um conjunto de escalas horizontais, o nível e declive de sincronismo e o modo de acoplamento com o circuito sob análise. Arquitecturalmente, o osciloscópio é constituído por um módulo especificado com a linguagem de descrição de hardware (HDL, Hardware Description Language) Verilog e por uma interface desenvolvida na linguagem de programação Java®. O módulo é embutido numa FPGA, definindo todo o processamento do osciloscópio. A interface permite o seu controlo e a representação do sinal medido. Durante o projecto foi utilizado um conversor Analógico/Digital (A/D) com uma frequência máxima de amostragem de 1.5MHz e 14 bits de resolução que, devido às suas limitações, obrigaram à implementação de um sistema de interpolação multi-estágio com filtros digitais.

Time series morphological analysis applied to biomedical signals events detection

Dissertation submitted in the fufillment of the requirements for the Degree of Master in Biomedical Engineering

Control system for actuation and sensing in digital microfluidics devices

Digital Microfluidics (DMF) is a second generation technique, derived from the conventional microfluidics that instead of using continuous liquid fluxes, it uses only individual droplets driven by external electric signals. In this thesis a new DMF control/sensing system for visualization, droplet control (movement, dispensing, merging and splitting) and real time impedance measurement have been developed. The software for the proposed system was implemented in MATLAB with a graphical user interface. An Arduino was used as control board and dedicated circuits for voltage switching and contacts were designed and implemented in printed circuit boards. A high resolution camera was integrated for visualization. In our new approach, the DMF chips are driven by a dual-tone signal where the sum of two independent ac signals (one for droplet operations and the other for impedance sensing) is applied to the electrodes, and afterwards independently evaluated by a lock-in amplifier. With this new approach we were able to choose the appropriated amplitudes and frequencies for the different proposes (actuation and sensing). The measurements made were used to evaluate the real time droplet impedance enabling the knowledge of its position and velocity. This new approach opens new possibilities for impedance sensing and feedback control in DMF devices.

The Gaussian assumption in second-order estimation problems in digital communications

This paper deals with the goodness of the Gaussian assumption when designing second-order blind estimationmethods in the context of digital communications. The low- andhigh-signal-to-noise ratio (SNR) asymptotic performance of the maximum likelihood estimator—derived assuming Gaussiantransmitted symbols—is compared with the performance of the optimal second-order estimator, which exploits the actualdistribution of the discrete constellation. The asymptotic study concludes that the Gaussian assumption leads to the optimalsecond-order solution if the SNR is very low or if the symbols belong to a multilevel constellation such as quadrature-amplitudemodulation (QAM) or amplitude-phase-shift keying (APSK). On the other hand, the Gaussian assumption can yield importantlosses at high SNR if the transmitted symbols are drawn from a constant modulus constellation such as phase-shift keying (PSK)or continuous-phase modulations (CPM). These conclusions are illustrated for the problem of direction-of-arrival (DOA) estimation of multiple digitally-modulated signals.

Statistical reference criteria for adaptive signal processing in digital communications

A general criterion for the design of adaptive systemsin digital communications called the statistical reference criterionis proposed. The criterion is based on imposition of the probabilitydensity function of the signal of interest at the outputof the adaptive system, with its application to the scenario ofhighly powerful interferers being the main focus of this paper.The knowledge of the pdf of the wanted signal is used as adiscriminator between signals so that interferers with differingdistributions are rejected by the algorithm. Its performance isstudied over a range of scenarios. Equations for gradient-basedcoefficient updates are derived, and the relationship with otherexisting algorithms like the minimum variance and the Wienercriterion are examined.

GPS-signal Source Based on Direct Digital Synthesis

This thesis presents the design and implementation of a GPS-signal source suitable for receiver measurements. The developed signal source is based on direct digital synthesis which generates the intermediate frequency. The intermediate frequency is transfered to the final frequency with the aid of an Inphase/Quadrature modulator. The modulating GPS-data was generated with MATLAB. The signal source was duplicated to form a multi channel source. It was shown that, GPS-signals ment for civil navigation are easy to generate in the laboratory. The hardware does not need to be technically advanced if navigation with high level of accuracy is not needed. It was also shown that, the Inphase/Quadrature modulator can function as a single side band upconverter even with a high intermediate frequency. This concept reduces the demands required for output filtering.