Music making and technological development has always been connected. The digital revolution has made advanced music production, writing and distribution tools universally accessible. New intelligent tools built on machine learning are entering the market potentially changing how we create music and interact with creative content.  The aim of this thesis project has been to find alternatives to existing interaction models manifested in modern DAWs (Digital Audio Workstations). Ideas developed through rough sketches and simple prototypes—the outcome consists of three concept videos proposing changes to three moments in the workflow of songwriters and producers. This thesis started with an idea of exploring the borderland between computer generated music and human creativity. Through desk research and interviews I learned that computational creativity exist and that there is a lot of different ways of defining creativity and art. Creating creative computers should not aim to replace humans creative abilities—it is rather about automating and creating tools that enhance our creative abilities.  To understand how songwriters and producers work the subject were investigated through semi-structured contextual interviews. The different ways of working and using tools were mapped out and potential opportunity areas were identified. This thesis have been a project that through sketching, mock-ups and simple prototypes questions how we use digital tools in music production. These concepts and sketches were continuously brought back to experts for feedback. The outcome consists of three concepts. They are presented through three short videos. These videos are now shared with a bigger audience and will act as an conversation starter for people interested in tools for digital music production.  1. Automating parts of the songwriting process and create a collaborative workflow between a you and a computer, through a conversational user interface.  2. A pressure sensitive touch surface that let you manipulate sound. It is an adaptive system that automatically detects active controls in your DAW—it maps these active controls from your computer screen down to a touch pad.  3. The third concept changes the way you organise and look for sound files. It is a automatised process where a software helps you compare different sounds to each other. It takes away most labels and focuses on mapping sound according to its auditory profile.

The aim of this work is to design an architecture of the embedded device that will be used for controlling DAW software in recording studio. First of all, attention is given to a brief summary of the necessary knowledge which is needed to design such kind of device. Af- ter that follows short survey of the existing solutions and description of protocols which can be used for communication with the recording software. Then, subsequent part of the thesis builds upon these foundations and further elaborates the device architecture by me- ans of decomposing it into several modules. In fact, two hardware modules are designed and manufactured, when each of them is conceived on a separate PCB with its own microcon- troller. Then the control firmware has been implemented for each of the modules. At the end of the work an aluminium enclosure, which holds both modules, is designed. The result of this work is a functional prototype of the assembled controller which can be used for the purpose of controlling DAW software.

The thesis deals with the analysis of technology used during the design of headphones with integrated active ambient noise cancellation and examines the possibilities of using adaptive filters to simplify development and achieve more effective attenuation.

In this work is presented technique for embedding digital watermark in digital audio signals. Digital watermark must be imperceptible and should be robust against attacks and other types of distortion. Algorithm is implemented for embedding digital watermark using technique spread-spectrum and psychoacoustic model ISO-MPEG I layer I. Robustness was tested for filtering signal, MP3 compression and resample method.

This diploma thesis deals with digital audio amplifier with universal inputs and its design. The first part describes modulation and audio formats for audio electronics. The thesis contain design of a block diagram of the digital audio amplifier and describes the requirements for functional blocks. As a basic device for audio signal processing was choosen integrated circuit STA326. The thesis continue with circuits design for each blocks with a description of their principles. The next section describes the construction and firmware for microcontroller. The last part of this diploma thesis is targeted on the presentation of the measured parameters of the amplifier. The conclusion summarizes the results that have been achieved and advantages and disadvantages of the digital audio amplifier prototype.

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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.

本研究針對我國學生學習日語發音進行相似度指標發展之初探，貢獻為針對目前日語發音提供一個相似度的指標可以和老師語音進行比較分析，找出分析日語發音相似度之模式。 研究從聲音數位化的角度切入，有別於過去研究使用語音辨識的方式來進行，聲音數位化後為數值的方式，因此使用指標來計算相似的程度。研究提出一套對應的聲音相似度指標，以電腦分析輔助日語學習者的發音練習。 指標建立過程由聲音取樣、正規化、端點偵測，到實際的運算，使用所蒐集的聲音資料來測試指標的穩定度與有效性，研究結果說明在以日語為母語者間的指標都很靠近，而不同日語腔調間會有一定的指標差異，對於一定日語程度的對象而言，指標落點很靠近，惟本研究此次蒐集到的聲音資料，其應用指標運算結果的分佈太過集中，如果能有更多樣化的聲音資料來測試指標應能有較漂亮的分佈圖形。 / This research includes developing a similarity index applies to the evaluation of Taiwan students learning Japanese pronunciation. The contribution of this research is that it provides a similarity index to the Japanese pronunciation comparing to the teacher’s pronunciation, finding the model of how to analysis the similarity of Japanese pronunciation. This research uses the digital audio processing to begin with, which is different from the other research that uses the speech recognition to evaluate the pronunciation. The audio will turn into numerical format after digitalize, so this research uses an index to calculate the similarity. By using this similarity index, the computer can become an assistant role that helps to analysis while learning Japanese pronunciation. The developing of index starts from audio sampling, audio normalizing, and end-point detection to the calculation of similarity index. This research collects audio data to test the stability and the validity of the similarity index. The result indicates that the similarity index of native Japanese speakers is very close；and the similarity index contains certain difference between different accents. For those Taiwan students who qualify with Japanese, their similarity index is close. Nevertheless, the result of the similarity index is too centralized, it would be better if there are more audio data to test the similarity index.

Ce mémoire de thèse de doctorat présente, discute et propose des outils de fouille automatique de mégadonnées dans un contexte de classification supervisée musical.L'application principale concerne la classification automatique des thèmes musicaux afin de générer des listes de lecture thématiques.Le premier chapitre introduit les différents contextes et concepts autour des mégadonnées musicales et de leur consommation.Le deuxième chapitre s'attelle à la description des bases de données musicales existantes dans le cadre d'expériences académiques d'analyse audio.Ce chapitre introduit notamment les problématiques concernant la variété et les proportions inégales des thèmes contenus dans une base, qui demeurent complexes à prendre en compte dans une classification supervisée.Le troisième chapitre explique l'importance de l'extraction et du développement de caractéristiques audio et musicales pertinentes afin de mieux décrire le contenu des éléments contenus dans ces bases de données.Ce chapitre explique plusieurs phénomènes psychoacoustiques et utilise des techniques de traitement du signal sonore afin de calculer des caractéristiques audio.De nouvelles méthodes d'agrégation de caractéristiques audio locales sont proposées afin d'améliorer la classification des morceaux.Le quatrième chapitre décrit l'utilisation des caractéristiques musicales extraites afin de trier les morceaux par thèmes et donc de permettre les recommandations musicales et la génération automatique de listes de lecture thématiques homogènes.Cette partie implique l'utilisation d'algorithmes d'apprentissage automatique afin de réaliser des tâches de classification musicale.Les contributions de ce mémoire sont résumées dans le cinquième chapitre qui propose également des perspectives de recherche dans l'apprentissage automatique et l'extraction de caractéristiques audio multi-échelles. / This doctoral dissertation presents, discusses and proposes tools for the automatic information retrieval in big musical databases.The main application is the supervised classification of musical themes to generate thematic playlists.The first chapter introduces the different contexts and concepts around big musical databases and their consumption.The second chapter focuses on the description of existing music databases as part of academic experiments in audio analysis.This chapter notably introduces issues concerning the variety and unequal proportions of the themes contained in a database, which remain complex to take into account in supervised classification.The third chapter explains the importance of extracting and developing relevant audio features in order to better describe the content of music tracks in these databases.This chapter explains several psychoacoustic phenomena and uses sound signal processing techniques to compute audio features.New methods of aggregating local audio features are proposed to improve song classification.The fourth chapter describes the use of the extracted audio features in order to sort the songs by themes and thus to allow the musical recommendations and the automatic generation of homogeneous thematic playlists.This part involves the use of machine learning algorithms to perform music classification tasks.The contributions of this dissertation are summarized in the fifth chapter which also proposes research perspectives in machine learning and extraction of multi-scale audio features.