En traitement du signal audio, l’analyse est une étape essentielle permettant de comprendre et d’inter-agir avec les signaux existants. En effet, la qualité des signaux obtenus par transformation ou par synthèse des paramètres estimés dépend de la précision des estimateurs utilisés. Cependant, des limitations théoriques existent et démontrent que la qualité maximale pouvant être atteinte avec une approche classique peut s’avérer insuffisante dans les applications les plus exigeantes (e.g. écoute active de la musique). Le travail présenté dans cette thèse revisite certains problèmes d’analyse usuels tels que l’analyse spectrale, la transcription automatique et la séparation de sources en utilisant une approche dite “informée”. Cette nouvelle approche exploite la configuration des studios de musique actuels qui maitrisent la chaîne de traitement avant l’étape de création du mélange. Dans les solutions proposées, de l’information complémentaire minimale calculée est transmise en même temps que le signal de mélange afin de permettre certaines transformations sur celui-ci tout en garantissant le niveau de qualité. Lorsqu’une compatibilité avec les formats audio existants est nécessaire, cette information est cachée à l’intérieur du mélange lui-même de manière inaudible grâce au tatouage audionumérique. Ce travail de thèse présente de nombreux aspects théoriques et pratiques dans lesquels nous montrons que la combinaison d’un estimateur avec de l’information complémentaire permet d’améliorer les performances des approches usuelles telles que l’estimation non informée ou le codage pur. / In the field of audio signal processing, analysis is an essential step which allows interactions with existing signals. In fact, the quality of transformed or synthesized audio signals depends on the accuracy over the estimated model parameters. However, theoretical limits exist and show that the best accuracy which can be reached by a classic estimator can be insufficient for the most demanding applications (e.g. active listening of music). The work which is developed in this thesis revisits well known audio analysis problems like spectral analysis, automatic transcription of music and audio sources separation using the novel ``informed'' approach. This approach takes advantage of a specific configuration where the parameters of the elementary signals which compose a mixture are known before the mixing process. Using the tools which are proposed in this thesis, the minimal side information is computed and transmitted with the mixture signal. This allows any kind of transformation of the mixture signal with a constraint over the resulting quality. When the compatibility with existing audio formats is required, the side information is embedded directly into the analyzed audio signal using a watermarking technique. This work describes several theoretical and practical aspects of audio signal processing. We show that a classic estimator combined with the sufficient side information can obtain better performances than classic approaches (classic estimation or pure coding).

In this thesis, we analyze the bandwidth requirements of MPEG-4 AVC video traffic and then propose and evaluate the accuracy of new MPEG-4 AVC video traffic models. First, we analyze the bandwidth requirements of the videos by comparing the statistical characteristics of the different frame types. We analyze their coefficient of variability, autocorrelation, and crosscorrelation in both short and long term. The Hurst parameter is also used to investigate the long range dependence of the video traces. We then provide an insight into B-frame dropping and its impact on the statistical characteristics of the video trace. This leads us to design two algorithms that predict the size of the B-frame and the size of the group of pictures (GOP) in the short-term. To evaluate the accuracy of the prediction, a model for the error is proposed. In a broadband cable network, B-frame size prediction can be employed by a cable headend to provision video bandwidth efficiently or more importantly, reduce bit rate variability and bandwidth requirements via selective B-frame dropping, thereby minimizing buffering requirements and packet losses at the set top box. It will be shown that the model provides highly accurate prediction, in particular for movies encoded in high quality resolution. The GOP size prediction can be used to provision bandwidth. We then enhance the B-frame and GOP size prediction models using a new scene change detector metric. Finally, we design an algorithm that predicts the size of different frame types in the long-term. Clearly, a long-term prediction algorithm may suffer degraded prediction accuracy and the higher complexity may result in higher latency. However, this is offset by the additional time available for long-term prediction and the need to forecast bandwidth usage well ahead of time in order to minimize packet losses during periods of peak bandwidth demands. We also analyze the impact of the video quality and the video standard on the accuracy of the model.

Orientador: Jose Eduardo Ribeiro de Paiva / Dissertação (mestrado) - Universidade Estadual de Campinas, Instituto de Artes / Made available in DSpace on 2018-08-11T16:40:10Z (GMT). No. of bitstreams: 1 Tonelli_MarcioJose_M.pdf: 9221864 bytes, checksum: ef171c3087236cc6e9b1afc9f845ce22 (MD5) Previous issue date: 2007 / Resumo: Este trabalho tem como objetivo principal discutir o impacto das redes móveis de comunicação nas atividades de criação, produção, divulgação, distribuição e acesso de música digital pelos usuários, considerando-se a tendência predominante no século 21 de criação de conteúdo informacional e de entretenimento por parte dos próprios usuários do ciberespaço. São analisados fenômenos como redes sociais e comunidades virtuais, nomadismo e tribalização, cibercultura e ciberespaço, interatividade, podeasting e a transformação do telefone celular numa nova mídia. Adicionalmente foi apresentado o aplicativo MobiDJ de composição de tones polifônicos, como uma ferramenta alinhada com a filosofia de conteúdo gerado pelo usuário (UGC - User Generated Content) e desenvolvida como um estudo de caso para embasamento deste trabalho. / Abstract:This dissertation has as its main objéetive the diseussion of the impaet of mobile eommunieations networks on the aetivities of ereation, produetion, promotion, distribution and digital musie aeeess by users taking into eonsideration the main trend of the 21 eentury of ereation of information and entertainment eontent by the end-users of eyberspaee themselves. Phenomena sueh as social networks and virtual eommunities, nomadism e tribal groups, eyber eulture and eyberspaee, interaetivity, podeasting and the transformation of eellular phones into a new media are analyzed. In addition the applieation software MobiDJ, an online platform for polyphonie tone eomposition was presented as a tool aligned with the philosophy of eontent generated by the end-user (UGC - User Generated Content) and developed as a ease study as rationale for this dissertation. / Mestrado / Mestre em Multimeios

L'objectif de cette thèse est d'étudier l'idée du tatouage dans le traitement du son.Les recherches en tatouage audio se sont principalement tournées vers des applications sécuritaires ou de transmission de données auxiliaires. Une des applications visées par ce concept consiste à améliorer la qualité du signal hôte ayant subi des transformations et ceci en exploitant l'information qu'il véhicule. Le tatouage audio est donc considéré comme mémoire porteuse d'informations sur le signal originel. La compression à bas débit des signaux audio est une des applications visée par ce concept. Dans ce cadre, deux objectifs sont proposés : la réduction du pré-écho et de l'amollissement d'attaque, deux phénomènes introduits par les codeurs audio perceptifs, en particulier les codeurs AAC et MP3; la préservation de l'harmonicité des signaux audio dégradée par les codeurs perceptifs à extension de bande, en particulier le codeur HE-AAC.La première partie de ce manuscrit présente les principes de base des systèmes de codage bas débit et étudie les différentes distorsions introduites par ces derniers. Fondées sur cette étude, deux solutions sont proposées. La première, visant principalement la réduction du pré-écho, consiste à corriger l'enveloppe temporelle du signal après réception en exploitant la connaissance a priori de l'enveloppe temporelle du signal original, supposée transmise par un canal auxiliaire à faible débit (< 500 bits/s). La seconde solution vise à corriger les ruptures d'harmonicité générées par les codeurs à extension de bande. Ce phénomène touche essentiellement les signaux fortement harmoniques (exemple : violon) et est perçu comme une dissonance. Une préservation de l'harmonicité des signaux audio par des opérations de translation spectrale est alors proposée, les paramètres étant là encore transmis par un canal auxiliaire à faible débit.La seconde partie de ce document est consacrée à l'intégration du tatouage audio dans les techniques de renforcement de la qualité des signaux audio précitées. Dans ce contexte, le tatouage audio remplace le canal auxiliaire précédent et œuvre comme une mémoire du signal originel, porteuse d'informations nécessaires pour la correction d'harmonicité et la réduction de pré-écho. Cette seconde partie a été précédée par une étape approfondie de l'évaluation des performances de la technique de tatouage adoptée en terme de robustesse à la compression MPEG (MP3, AAC et aacPlus). / The goal of this thesis is to explore the idea of watermark for sound enhancement. Classically, watermark schemes are oriented towards security applications or maximization of the transmitted bit rates. Our approach is completely different. Our goal is to study how an audio watermarking can improve the quality of the host audio signal by exploiting the information it conveys. The audio watermarking is considered as a memory that carries information about the original signal.The low bitrate compression of audio signals is one of the applications covered by this concept. In this context, two objectives are proposed: reducing the pre-echo and the attack softening, two phenomena introduced by the perceptual audio coders, particularly AAC and MP3 encoders ; preserving the harmonicity of audio signals, distorted by coders with bandwidth extension, especially HE-AAC encoder. These coders are limited in the reconstruction of the high-frequency spectrum mainly because of the potential unpredictability of the fine structure of the latter, as well as imperfect indicators of tonal to noise.The first part of this manuscript presents the basic principles of low rate coding systems and studies the various distortions introduced by the latter. Based on this study, two solutions are proposed. The first one, principally aimed at reducing the pre-echo, consist in correcting the time envelope of the signal after reception by exploiting the prior knowledge of the temporal envelope of the original signal, which is assumed transmitted by an auxiliary channel at low bitrates (<500 bps). The second solution is to correct the harmonicity generated by coders with bandwidth extension. This primarily affects strongly harmonic signals (e.g. violin) and is perceived as a dissonance. We propose then to preserve the harmonicity of audio signals by spectral translations. The parameters being passed again by an auxiliary channel at low bitrates.The second part of this document is dedicated to the integration of audio watermarking techniques in the solution presented in the first part. In this context, the audio watermarking replaces the previous auxiliary channel and is regarded as a memory of the original signal, carrying information necessary for the correction of harmonicity and the pre-echo reduction.

Résumé : Afin de répondre aux besoins de communication et de divertissement, il ne fait aucun doute que la parole et l’audio doivent être encodés sous forme numérique. En qualité CD, cela nécessite un débit numérique de 1411.2 kb/s pour un signal stéréo-phonique. Une telle quantité de données devient rapidement prohibitive pour le stockage de longues durées d’audio ou pour la transmission sur certains réseaux, particulièrement en temps réel (d’où l’adhésion universelle au format MP3). De plus, ces dernières années, la quantité de productions musicales et cinématographiques disponibles en cinq canaux et plus ne cesse d’augmenter. Afin de maintenir le débit numérique à un niveau acceptable pour une application donnée, il est donc naturel pour un codeur audio à bas débit d’exploiter la redondance entre les canaux et la psychoacoustique binaurale. Le codage perceptuel et plus particulièrement le codage paramétrique permet d’atteindre des débits manifestement inférieurs en exploitant les limites de l’audition humaine (étudiées en psychoacoustique). Cette recherche se concentre donc sur le codage paramétrique à bas débit de plus d’un canal audio. // Abstract : In order to fulfill our communications and entertainment needs, there is no doubt that speech and audio must be encoded in digital format. In"CD" quality, this requires a bit-rate of 1411.2 kb/s for a stereo signal. Such a large amount of data quickly becomes prohibitive for long-term storage of audio or for transmitting on some networks, especially in real-time (leading to a universal adhesion to the MP3 format). Moreover, throughout the course of these last years, the number of musical and cinematographic productions available in five channels or more continually increased.In order to maintain an acceptable bit-rate for any given application, it is obvious that a low bit-rate audio coder must exploit the redundancies between audio channels and binaural psychoacoustics. Perceptual audio coding, and more specifically parametric audio coding, offers the possibility of achieving much lower bit-rates by taking into account the limits of human hearing (psychoacoustics). Therefore, this research concentrates on parametric audio coding of more than one audio channel.

Résumé : L’audio numérique s’est déployé de façon phénoménale au cours des dernières décennies, notamment grâce à l’établissement de standards internationaux. En revanche, l’imposition de normes introduit forcément une certaine rigidité qui peut constituer un frein à l’amélioration des technologies déjà déployées et pousser vers une multiplication de nouveaux standards. Cette thèse établit que les codecs existants peuvent être davantage valorisés en améliorant leur qualité ou leur débit, même à l’intérieur du cadre rigide posé par les standards établis. Trois volets sont étudiés, soit le rehaussement à l’encodeur, au décodeur et au niveau du train binaire. Dans tous les cas, la compatibilité est préservée avec les éléments existants. Ainsi, il est démontré que le signal audio peut être amélioré au décodeur sans transmettre de nouvelles informations, qu’un encodeur peut produire un signal amélioré sans ajout au décodeur et qu’un train binaire peut être mieux optimisé pour une nouvelle application. En particulier, cette thèse démontre que même un standard déployé depuis plusieurs décennies comme le G.711 a le potentiel d’être significativement amélioré à postériori, servant même de cœur à un nouveau standard de codage par couches qui devait préserver cette compatibilité. Ensuite, les travaux menés mettent en lumière que la qualité subjective et même objective d’un décodeur AAC (Advanced Audio Coding) peut être améliorée sans l’ajout d’information supplémentaire de la part de l’encodeur. Ces résultats ouvrent la voie à davantage de recherches sur les traitements qui exploitent une connaissance des limites des modèles de codage employés. Enfin, cette thèse établit que le train binaire à débit fixe de l’AMR WB+ (Extended Adaptive Multi-Rate Wideband) peut être compressé davantage pour le cas des applications à débit variable. Cela démontre qu’il est profitable d’adapter un codec au contexte dans lequel il est employé. / Abstract : Digital audio applications have grown exponentially during the last decades, in good part because of the establishment of international standards. However, imposing such norms necessarily introduces hurdles that can impede the improvement of technologies that have already been deployed, potentially leading to a proliferation of new standards. This thesis shows that existent coders can be better exploited by improving their quality or their bitrate, even within the rigid constraints posed by established standards. Three aspects are studied, being the enhancement of the encoder, the decoder and the bit stream. In every case, the compatibility with the other elements of the existent coder is maintained. Thus, it is shown that the audio signal can be improved at the decoder without transmitting new information, that an encoder can produce an improved signal without modifying its decoder, and that a bit stream can be optimized for a new application. In particular, this thesis shows that even a standard like G.711, which has been deployed for decades, has the potential to be significantly improved after the fact. This contribution has even served as the core for a new standard embedded coder that had to maintain that compatibility. It is also shown that the subjective and objective audio quality of the AAC (Advanced Audio Coding) decoder can be improved, without adding any extra information from the encoder, by better exploiting the knowledge of the coder model’s limitations. Finally, it is shown that the fixed rate bit stream of the AMR-WB+ (Extended Adaptive Multi-Rate Wideband) can be compressed more efficiently when considering a variable bit rate scenario, showing the need to adapt a coder to its use case.

Submitted by William Justo Figueiro (williamjf) on 2015-07-08T20:56:12Z No. of bitstreams: 1 03b.pdf: 3126214 bytes, checksum: 0f98dbf86ae74816af91944aa7dec80f (MD5) / Made available in DSpace on 2015-07-08T20:56:12Z (GMT). No. of bitstreams: 1 03b.pdf: 3126214 bytes, checksum: 0f98dbf86ae74816af91944aa7dec80f (MD5) Previous issue date: 2013 / Nenhuma / A qualidade perceptual alcançada pelos codificadores de áudio depende diretamente da escolha de seus parâmetros. O codificador MPEG-4 AAC (Advanced Audio Coding), utilizado no Sistema Brasileiro de Televisão Digital (SBTVD), possui em sua estrutura uma etapa composta por um laço de iteração para escolher os parâmetros do codificador, de maneira dinâmica durante o processo de codificação. Este processo de escolha pode ser definido como um problema de Pesquisa Operacional, sendo um problema de Seleção de Partes, denominado como o Problema de Codificação AAC. A estrutura existente no codificador de referência, não resolve este problema de maneira ótima. Desta forma, este trabalho propõe o desenvolvimento e implementação de um modelo de uma estrutura de simulação, para encontrar os parâmetros do codificador de áudio MPEG-4 AAC, de maneira a otimizar a qualidade perceptual do áudio, para uma determinada taxa de bits (bit rate). A implementação da estrutura de otimização foi desenvolvida em linguagem C, utilizando as metaheurísticas Busca Tabu e Algoritmo Genético em uma estrutura híbrida. Através da minimização da métrica ANMR (Average Noise-to-Mask Ratio), o algoritmo procura identificar a melhor configuração dos parâmetros internos do codificador MPEG-4 AAC, de maneira que possa garantir uma qualidade perceptual para o sinal áudio. Os resultados obtidos utilizando a estrutura híbrida de otimização apresentaram valores menores para a métrica ANMR, ou seja, uma melhor qualidade perceptual de áudio, quando comparados com os resultados obtidos com o codificador de referência MPEG-4 AAC. / The perceptual quality achieved by audio encoders depends directly on the choice of its parameters. The MPEG-4 AAC (Advanced Audio Coding), used in the Brazilian Digital Television System (BDTS), has a step in its structure that consists in iteration loop to choose the parameters of the encoder dynamically during the encoding process. This selection process can be defined as a problem of Operational Research, being a Part Selection Problem, termed as AAC Encoding Problem. The structure in the reference encoder not solves this problem optimally. Thus, this paper proposes the development and implementation of a model simulation of a structure, to find the internal parameters of the MPEG-4 AAC audio encoder, so as to optimize the perceptual audio quality for a given bit rate. The implementation of the optimization framework was developed in ANSI C programming language, using the Tabu Search and Genetic Algorithm metaheuristics in a hybrid structure. Through the minimization of the ANMR (Average Noise-to-Mask Ratio) metric, the algorithm tries to identify the best configuration of internal parameters of the MPEG-4 AAC. The results obtained using the optimization hybrid structure achieve lower values for the ANMR metric, i.e., an better perceptual audio quality, compared with the obtained with the reference encoder MPEG-4 AAC.

MPEG audio compression techniques significantly reduces the storage and transmission requirements for high quality digital audio. However, compression complicates the processing of audio in many applications. If a compressed audio signal is to be processed, a direct method would be to decode the compressed signal, process the decoded signal and re-encode it. This is computationally expensive due to the complexity of the MPEG filter bank. This thesis deals with processing of MPEG compressed audio. The main contributions of this thesis are a) Extracting wavelet coefficients in the MPEG compressed domain. b) Wavelet based pitch extraction in MPEG compressed domain. c) Time Scale Modifications of MPEG audio. d) Watermarking of MPEG audio. The research contributions starts with a technique for calculating several levels of wavelet coefficients from the output of the MPEG analysis filter bank. The technique exploits the toeplitz structure which arises when the MPEG and wavelet filter banks are represented in a matrix form, The computational complexity for extracting several levels of wavelet coefficients after decoding the compressed signal and directly from the output of the MPEG analysis filter bank are compared. The proposed technique is found to be computationally efficient for extracting higher levels of wavelet coefficients. Extracting pitch in the compressed domain becomes essential when large multimedia databases need to be indexed. For example one may be interested in listening to a particular speaker or to listen to male female audio segments in a multimedia document. For this application, pitch information is one of the very basic and important features required. Pitch is basically the time interval between two successive glottal closures. Glottal closures are accompanied by sharp transients in the speech signal which in turn gives rise to a local maxima in the wavelet coefficients. Pitch can be calculated by finding the time interval between two successive maxima in the wavelet coefficients. It is shown that the computational complexity for extracting pitch in the compressed domain is less than 7% of the uncompressed domain processing. An algorithm for extracting pitch in the compressed domain is proposed. The result of this algorithm for synthetic signals, and utterances of words by male/female is reported. In a number of important applications, one needs to modify an audio signal to render it more useful than its original. Typical applications include changing the time evolution of an audio signal (increase or decrease the rate of articulation of a speaker),or to adapt a given audio sequence to a given video sequence. In this thesis, time scale modifications are obtained in the subband domain such that when the modified subband signals are given to the MPEG synthesis filter bank, the desired time scale modification of the decoded signal is achieved. This is done by making use of sinusoidal modeling [I]. Here, each of the subband signal is modeled in terms of parameters such as amplitude phase and frequencies and are subsequently synthesised by using these parameters with Ls = k La where Ls is the length of the synthesis window , k is the time scale factor and La is the length of the analysis window. As the PCM version of the time scaled signal is not available, psychoacoustic model based bit allocation cannot be used. Hence a new bit allocation is done by using a subband coding algorithm. This method has been satisfactorily tested for time scale expansion and compression of speech and music signals. The recent growth of multimedia systems has increased the need for protecting digital media. Digital watermarking has been proposed as a method for protecting digital documents. The watermark needs to be added to the signal in such a way that it does not cause audible distortions. However the idea behind the lossy MPEC encoders is to remove or make insignificant those portions of the signal which does not affect human hearing. This renders the watermark insignificant and hence proving ownership of the signal becomes difficult when an audio signal is compressed. The existing compressed domain methods merely change the bits or the scale factors according to a key. Though simple, these methods are not robust to attacks. Further these methods require original signal to be available in the verification process. In this thesis we propose a watermarking method based on spread spectrum technique which does not require original signal during the verification process. It is also shown to be more robust than the existing methods. In our method the watermark is spread across many subband samples. Here two factors need to be considered, a) the watermark is to be embedded only in those subbands which will make the addition of the noise inaudible. b) The watermark should be added to those subbands which has sufficient bit allocation so that the watermark does not become insignificant due to lack of bit allocation. Embedding the watermark in the lower subbands would cause distortion and in the higher subbands would prove futile as the bit allocation in these subbands are practically zero. Considering a11 these factors, one can introduce noise to samples across many frames corresponding to subbands 4 to 8. In the verification process, it is sufficient to have the key/code and the possibly attacked signal. This method has been satisfactorily tested for robustness to scalefactor, LSB change and MPEG decoding and re-encoding.

This dissertation explores Mobile Media Technologies (MMT’s) namely, cellphones, laptops and MP3 players, and their prevalence in public space as well as how they are being used within the space. Much of my research analyses the impact of MMT’s on social behaviour and the extent to which they can be seen as the harbingers of a new “postmodern” form of social organisation. My research is predominantly an observational study which is conducted within the postmodern space of the shopping mall. Through my research I discuss the multiple spaces within the shopping mall environment and I explore how humans behave, interact and construct their identities within this space; these ideas are evaluated in terms of the “modern” and the “postmodern” paradigms. “Postmodernity” and “modernity” are not mutually exclusive and as a result there are ambivalences in terms of how individuals relate to how MMT’s are being used in public space. / Thesis (M.A.)-University of KwaZulu-Natal, Pietermaritzburg, 2008.

This work deals with the problem of reverse audio engineering for active listening. The format under consideration corresponds to the audio CD. The musical content is viewed as the result of a concatenation of the composition, the recording, the mixing, and the mastering. The inversion of the two latter stages constitutes the core of the problem at hand. The audio signal is treated as a post-nonlinear mixture. Thus, the mixture is "decompressed" before being "decomposed" into audio tracks. The problem is tackled in an informed context: The inversion is accompanied by information which is specific to the content production. In this manner, the quality of the inversion is significantly improved. The information is reduced in size by the use of quantification and coding methods, and some facts on psychoacoustics. The proposed methods are applicable in real time and have a low complexity. The obtained results advance the state of the art and contribute new insights.