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.

TV Digital aberta no Brasil

O presente trabalho analisa os importantes desafios que as novas tecnologias e as transformações na sociedade pós-industrial estão impondo à TV digital brasileira e ao seu modelo de negócios. Tem como principal objetivo realizar uma reflexão sobre a viabilidade financeira das emissoras abertas com a chegada da TV digital. Para tanto, analisa o modelo de negócios anterior, da TV analógica, baseado nos comerciais de trinta segundos e como esta forma poderá ser afetada inviabilizando a estrutura de produção e distribuição de conteúdo pelas emissoras abertas digitais. Ainda, busca evidenciar demais fatores que contribuem para a migração da audiência para outras plataformas de distribuição de conteúdo, refutando o senso comum de que o acesso à internet banda larga é a principal causa da queda dos índices de audiência. Este estudo se utiliza de uma ampla bibliografia que extrapola o campo específico da Comunicação e amplia o olhar sobre a indústria televisiva aberta no Brasil, enumerando fragilidades do setor e apontando possíveis estratégias para que a televisão brasileira possa se adaptar à nova estrutura da comunicação que está se formando em nosso país.

Musical genres: beating to the rhythms of different drums

Online music databases have increased significantly as a consequence of the rapid growth of the Internet and digital audio, requiring the development of faster and more efficient tools for music content analysis. Musical genres are widely used to organize music collections. In this paper, the problem of automatic single and multi-label music genre classification is addressed by exploring rhythm-based features obtained from a respective complex network representation. A Markov model is built in order to analyse the temporal sequence of rhythmic notation events. Feature analysis is performed by using two multi-variate statistical approaches: principal components analysis (unsupervised) and linear discriminant analysis (supervised). Similarly, two classifiers are applied in order to identify the category of rhythms: parametric Bayesian classifier under the Gaussian hypothesis (supervised) and agglomerative hierarchical clustering (unsupervised). Qualitative results obtained by using the kappa coefficient and the obtained clusters corroborated the effectiveness of the proposed method.

Comparação da implementação da rede TDT em frequência única e multifrequência

Trabalho Final de Mestrado para obtenção do grau de Mestre em Engenharia de Electrónica e Telecomunicações

Automatic DVB signal analyser

El problema de controlar les emissions de televisió digital a tota Europa pel desenvolupament de receptors robustos i fiables és cada vegada més significant, per això, sorgeix la necessitat d’automatitzar el procés d’anàlisi i control d’aquests senyals. Aquest projecte presenta el desenvolupament software d’una aplicació que vol solucionar una part d’aquest problema. L’aplicació s’encarrega d’analitzar, gestionar i capturar senyals de televisió digital. Aquest document fa una introducció a la matèria central que és la televisió digital i la informació que porten els senyals de televisió, concretament, la que es refereix a l’estàndard "Digital Video Broadcasting". A continuació d’aquesta part, l’escrit es concentra en l’explicació i descripció de les funcionalitats que necessita cobrir l'aplicació, així com introduir i explicar cada etapa d’un procés de desenvolupament software. Finalment, es resumeixen els avantatges de la creació d’aquest programa per l’automatització de l’anàlisi de senyal digital partint d’una optimització de recursos.

Inserció de marques inaudibles en ones d'àudio

Es vol implementar un sistema de detecció de còpia per a protegir el copyright de fitxers d'àudio digitals en format WAV. El sistema haurà de contenir dos algorismes bàsics: un per a inserir una marca d'aigua (

Disseny d'un sistema de filtrat d'àudio

Tant el medi transmissor com els equips d'enregistrament o reproducció de so introdueixen components de soroll d'alta freqüència als senyals. En aquest treball de final de carrera (TFC), s'ha dissenyat i implementat un sistema de filtrat d'àudio encaminat a filtrar aquestes components d'alta freqüència. Donat que l'oïda humana no pot percebre sons de més de 20 kHz, s'ha considerat aquest límit com a freqüència màxima a mantenir en la senyal.S'ha començat estudiant el senyal problema a través del seu espectre de freqüències simulat mitjançant la transformada discreta de Fourier (DFT, en anglès). Una vegada identificades les components d'alta freqüència a atenuar, s'han estudiat les diferents opcions de filtre passabaix.Inicialment, s'ha valorat la possibilitat del disseny de filtres analògics de Butterworth o Chebyshev, o de filtres digitals de tipus IIR (Infinite Impulse Response) basats en els primers. Tanmateix, malgrat assolir les especificacions en magnitud, mitjançant aquest filtres no s'obté una fase lineal en la banda de pas. Per això, s'ha realitzat un disseny de filtre digital tipus FIR (Finite Infinite Response) que compleix estrictament amb les especificacions i presenta una fase lineal en la banda de pas. S'ha simulat el comportament d'aquest filtre amb el senyal problema per tal d'assegurar el seu correcte funcionament.A continuació, s'ha implementat aquest últim disseny en llenguatge C i compilat per un microcontrolador de l'empresa Microchip. S'han realitzat proves de simulació mitjançant Stimulus del programa MPLAB. En definitiva, s'ha dissenyat un filtre passabaix de tipus FIR per acondicionar una senyal d'àudio que posteriorment s'ha implementat en un microcontrolador de Microchip.

Le régime de la copie privée face à Internet

Le présent article vise à exposer la situation du régime de la copie privée face à Internet, tant dans son application que dans sa légitimité. L’auteur soumet notamment que l’exception de la copie privée n’est applicable ni à la mise à disposition ni au téléchargement de fichiers musicaux sur Internet à moins que, quant à ce dernier acte, la copie provienne d’une source licite, ce qui exclut les réseaux peer-to-peer. Ensuite, l’auteur analyse la question quant à savoir si les enregistreurs audionumériques et les disques durs des ordinateurs personnels devraient être soumis aux redevances au Canada, et conclut à l’effet qu’une réforme de la Loi sur le droit d’auteur est de mise si l’on veut en arriver à une telle éventualité. Finalement dans la dernière partie, l’auteur soulève la question de la remise en cause du régime de la copie privée face à l’importance grandissante que l‘on accorde à la gestion numérique des droits.

Centor : concept d’interface de spatialisation sonore additive

Cette recherche porte un regard critique sur les interfaces de spatialisation sonore et positionne la composition de musique spatiale, un champ d’étude en musique, à l’avant plan d’une recherche en design. Il détaille l’approche de recherche qui est centrée sur le processus de composition de musique spatiale et les modèles mentaux de compositeurs électroacoustiques afin de livrer des recommandations de design pour le développement d’une interface de spatialisation musicale nommée Centor. Cette recherche montre qu’un processus de design mené à l’intersection du design d’interface, du design d’interaction et de la théorie musicale peut mener à une proposition pertinente et innovatrice pour chacun des domaines d’étude. Nous présentons la recherche et le développement du concept de spatialisation additive, une méthode de spatialisation sonore par patrons qui applique le vocabulaire spectromorphologique de Denis Smalley. C’est un concept d’outil de spatialisation pour le studio qui complémente les interfaces de composition actuelles et ouvre un nouveau champ de possibilités pour l’exploration spatiale en musique électroacoustique. La démarche de recherche présentée ici se veut une contribution au domaine du design d’interfaces musicales, spécifiquement les interfaces de spatialisation, mais propose aussi un processus de design pour la création d’interfaces numériques d’expression artistique.

Efficient Watermarking Schemes for Speaker Verification Guaranteeing Non-repudiation

Presently different audio watermarking methods are available; most of them inclined towards copyright protection and copy protection. This is the key motive for the notion to develop a speaker verification scheme that guar- antees non-repudiation services and the thesis is its outcome. The research presented in this thesis scrutinizes the field of audio water- marking and the outcome is a speaker verification scheme that is proficient in addressing issues allied to non-repudiation to a great extent. This work aimed in developing novel audio watermarking schemes utilizing the fun- damental ideas of Fast-Fourier Transform (FFT) or Fast Walsh-Hadamard Transform (FWHT). The Mel-Frequency Cepstral Coefficients (MFCC) the best parametric representation of the acoustic signals along with few other key acoustic characteristics is employed in crafting of new schemes. The au- dio watermark created is entirely dependent to the acoustic features, hence named as FeatureMark and is crucial in this work. In any watermarking scheme, the quality of the extracted watermark de- pends exclusively on the pre-processing action and in this work framing and windowing techniques are involved. The theme non-repudiation provides immense significance in the audio watermarking schemes proposed in this work. Modification of the signal spectrum is achieved in a variety of ways by selecting appropriate FFT/FWHT coefficients and the watermarking schemes were evaluated for imperceptibility, robustness and capacity char- acteristics. The proposed schemes are unequivocally effective in terms of maintaining the sound quality, retrieving the embedded FeatureMark and in terms of the capacity to hold the mark bits. Robust nature of these marking schemes is achieved with the help of syn- chronization codes such as Barker Code with FFT based FeatureMarking scheme and Walsh Code with FWHT based FeatureMarking scheme. An- other important feature associated with this scheme is the employment of an encryption scheme towards the preparation of its FeatureMark that scrambles the signal features that helps to keep the signal features unreve- laed. A comparative study with the existing watermarking schemes and the ex- periments to evaluate imperceptibility, robustness and capacity tests guar- antee that the proposed schemes can be baselined as efficient audio water- marking schemes. The four new digital audio watermarking algorithms in terms of their performance are remarkable thereby opening more opportu- nities for further research.

El control interno y su desarrollo normativo en Colombia

La modernización institucional puesto de manifiesto con la nueva Constitución toda vez que sus preceptos impone a los organismos estatales ponerse a la par en la estructura y organización con los nuevos métodos y sistemas de administración gerencial

A relative path attenuation approximation algorithm for DVB-T systems

In the United Kingdom and in fact throughout Europe, the chosen standard for digital terrestrial television is the European Telecommunications Standards Institute (ETSI) ETN 300 744 also known as Digital Video Broadcasting - Terrestrial (DVB-T). The modulation method under this standard was chosen to be Orthogonal Frequency Division Multiplex (0FD4 because of the apparent inherent capability for withstanding the effects of multipath. Within the DVB-T standard, the addition of pilot tones was included that can be used for many applications such as channel impulse response estimation or local oscillator phase and frequency offset estimation. This paper demonstrates a technique for an estimation of the relative path attenuation of a single multipath signal that can be used as a simple firmware update for a commercial set-top box. This technique can be used to help eliminate the effects of multipath(1).

An analysis on the performance of the forward and return interaction channels in DVB-RCS

The next generation consumer level interactive services require reliable and constant communication for both mobile and static users. The Digital Video Broadcasting ( DVB) group has exploited the rapidly increasing satellite technology for the provision of interactive services and launched a standard called Digital Video Broadcast through Return Channel Satellite (DYB-RCS). DVB-RCS relies on DVB-Satellite (DVB-S) for the provision of forward channel. The Digital Signal processing (DSP) implemented in the satellite channel adapter block of these standards use powerful channel coding and modulation techniques. The investigation is concentrated towards the Forward Error Correction (FEC) of the satellite channel adapter block, which will help in determining, how the technology copes with the varying channel conditions and user requirements(1).

Analysis of a DVB-T compliant receiver simulation under various multipath conditions

Since 1966, coded orthogonal frequency division multiplexing (COFDM) has been investigated to determine the possibility of reducing the overall throughput of a digitally modulated terrestrial television channel. In the investigations, many assumptions have emerged. One common misconception is that in a terrestrial environment, COFDM has an inherent immunity to multipath interference. A theoretical analysis of a multipath channel, along with simulation results has shown that this assumption does not hold the information is considered when including the radio frequency modulation and demodulation. This paper presents a background into the inception of COFDM, a mathematical analysis of the digitally modulated television signal under multipath conditions and the results of a European Digital Video Broadcasting-Terrestrial (DVB-T) compliant simulation model with MPEG-2 bitstreams transmitted under various multipath conditions.

Reconstructing The Incredible

An amplified scissors confronts its digital symbol in this audio piece created for RELAY, an online music project devised and curated by Irish musician John Lambert aka Chequerboard. Digital audio editing cuts are placed randomly over scissors sound samples and then performed with the scissors instrument to determine the rhythm of the composition. RELAY creates a chain of sound pieces where each work is created in response to the previous so that ideas and sounds shift, mutate and evolve over time. Commissioned by Model Arts and Niland Gallery, Sligo, (Ireland).