Electro-acoustic music composition makes it possible for composers to manipulate sounds with the computer with either alone or both live and prerecorded sounds. From the end of the 19th century when the first electronic devices for performing music were developed up till today, transformations in music technology keep surfacing and the possibilities with working with sounds have become endless. Electro-acoustic music has the ability to conjure mental images using vast sound manipulation techniques using the DAW(Digital audio workstation) and sound amplification through loudspeakers. Every space in our environment has its unique sound/s. Lost and found through these techniques provides an insight to a space which seems distant by geographical location but closer through sound.

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Dans des systèmes numériques essentiellement latents, compenser la latence n’est pastrivial, en particulier lorsque les graphes de routage du signal sont complexes commec’est souvent le cas dans une station audionumérique (DAW).Tandis que le problème général est de nature mathématique, des complicationsapparaissent dans la conception de systèmes audio en temps réel à cause des contraintesdu matériel, de l’architecture du système, ou de l’ingénierie.Pour construire un système fournissant une compensation de latence sur l’intégralitédu graphe avec possibilité de connecter n’importe quelle source à n’importe quelledestination, uniquement décrire les mécanismes est insuffisant. Le système completdoit être conçu d’un bloc à l’aide de prototypes pour prendre en compte les limitationsdu monde réel.Cette recherche a été menée en utilisant Ardour, une station audionumériquelibrement disponible sous licence libre GPL. Cette thèse est autant un rapport deconception qu’une documentation de recherche.Une analyse complète des éléments de base et de leurs interactions est présentée.La plupart ont été implémentés au delà de la démonstration de faisabilité, dans lebut de combler l’écart entre les systèmes professionnels de production audio et ladocumentation librement accessible pour la recherche et le développement.Même si elle s’attache ostensiblement à Ardour, cette thèse décrit les conceptgénériques des station audio tels que les Ports, les pistes (Tracks), les bus (Busses)et les processeurs de traitement numériques du signal (Processors) ainsi que lesinteractions opérationnelles entre eux.Les concepts de base communs à toutes les entrées/sorties numériques sont expliquésainsi que les sources de latence. Les graphes de traitement et de latence sont illustréspour présenter une vue d’ensemble.Les problèmes généraux rencontrés lors de l’alignement temporel, tant local que / In inherently latent digital systems it is not trivial to compensate for latency, particularlyin situations of complex signal routing graphs as is the case in a Digital AudioWorkstation.While the general problem is of mathematical nature, design complexities arisein real-time audio systems due to constraints by hardware, system-architecture andengineering.To construct a system providing for full-graph latency compensation with anywhereto-anywhere routing capabilities, it is insufficient to merely describe mechanisms.The complete system has to be designed as one and prototyped to take real-worldlimitations into account.This research was carried out using Ardour, a digital audio workstation, whichis freely available under the GPL free-software licence. This thesis is as much adesign-report as it is research documentation.A complete breakdown of building-blocks and interaction is presented, most of whichhas also been implemented beyond a proof-of-concept with the goal to bridge the gapbetween professional audio production systems and freely accessible documentationfor research and development.While ostensibly focusing on Ardour, this thesis describes generic concepts of AudioWorkstations like Ports, Tracks, Busses, and DSP Processors, as well as operationalinteraction between them.Basic concepts common to all digital I/O processes an,d sources of latency areexplained, and process- and latency graphs are illustrated to provide a completepicture. General issues related to time-alignment, both local, and global, as wellas more DAW specific cases like parameter-automation and parallel-execution arediscussed. Algorithms are modelled with pseudocode where appropriate and applicationprogramming interfaces are presented as examples to concepts throughout the text.

<p>This thesis describes the design and implementation of a Digital Audio Broadcasting (DAB) System developed using C++ Language and SystemC libraries. The main aspects covered within this report are the data structure of DAB system, and some interesting points of SystemC Library very useful for the implementation of the final system. </p><p>It starts with a introduction of DAB system and his principals advantages. Next it goes further into the definition of data structures of DAB, they are FIC, MSC, and DAB audio frame, explained with MPEG and PAD packets. Later on this chapter there is an explanation of the SystemC library with special attention on the features that I used to implement the system. This features are the events used in the communication between processes and the interfaces needed for sending and receiving the data.</p><p>With all these points covered is quite easy for a reader to understand the implementation of the system, despite this point is covered in the last chapter of the thesis. The implementation is here explained in two different steps. The first one explain how is formed the DAB audio frame by means of MPEG frames that are wrote in channel by producer interface, this frames are readed by consumer interface. For this purpose I have created some classes and structures that are explained in this part. The second part explain how I obtain the DAB transmission frame which is obtained creating MSC frames, that are big data structures formed by groups of DAB audio frames, therefore there are some functions that act like a buffer and add audio frames to the MSC data structure. Of independent way there is the FIC frame that is generated of random way and its added to the transmission frame.</p>

This thesis describes the design and implementation of a Digital Audio Broadcasting (DAB) System developed using C++ Language and SystemC libraries. The main aspects covered within this report are the data structure of DAB system, and some interesting points of SystemC Library very useful for the implementation of the final system. It starts with a introduction of DAB system and his principals advantages. Next it goes further into the definition of data structures of DAB, they are FIC, MSC, and DAB audio frame, explained with MPEG and PAD packets. Later on this chapter there is an explanation of the SystemC library with special attention on the features that I used to implement the system. This features are the events used in the communication between processes and the interfaces needed for sending and receiving the data. With all these points covered is quite easy for a reader to understand the implementation of the system, despite this point is covered in the last chapter of the thesis. The implementation is here explained in two different steps. The first one explain how is formed the DAB audio frame by means of MPEG frames that are wrote in channel by producer interface, this frames are readed by consumer interface. For this purpose I have created some classes and structures that are explained in this part. The second part explain how I obtain the DAB transmission frame which is obtained creating MSC frames, that are big data structures formed by groups of DAB audio frames, therefore there are some functions that act like a buffer and add audio frames to the MSC data structure. Of independent way there is the FIC frame that is generated of random way and its added to the transmission frame.

In this thesis, the author explored an alternative way of composing contemporary western art music, being inspired by thoughts regarding post-genre. The composition method incorporated the use of the Digital Audio Workstation (DAW) software Cubase. Musical gestures were recorded with two musicians which where used as samples in the DAW to compose the piece Lines. The role of the score was shifted by moving it’s realisation to after the aural result was completed. The process was inspiring and was perceived to work well for a textural piece of music. Using a DAW when composing contemporary western art music is something that could be explored by classically trained composers in favour of working solely in a notation software. The result was the digitally made recording of Lines and two scores; one aimed to reproduce the recorded version (where only violin and violoncello where used) and another where adjustments where made, mainly in the instrumentation, to facilitate live performances by string orchestras.

There is a lot of research connecting musical expression with well-being, motivation and meaningfulness, especially when introduced in early years. Children should have the opportunity to express themselves through music regardless of motor disability. There is research targeting children with motor disabilities and musical expression but when the projects ends the children are left without the technology. There is a gap in research exploring technologies accessible in terms of availability to buy and use for the children. This thesis take advantage of the evolution of music production technologies where standard digital audio workstations are highly customizable and therefore an option to use as tools to democratize musical expression among children with motor disabilities. Democratize in this context means to make the children able to participate in a musical performance and express themselves in a similar way as fully abled children. To explore the use of music production technologies to promote musical expression among children with motor disabilities a proposed solution of digital audio workstation Ableton Live and beat matched functionalities coupled with various tangible controllers was explored during 3 sessions at a school in Malmö. Four children at the age of 10 with motor disabilities participated together with their music teacher. The result indicates that the proposed solution has great potential to democratize musical expression among the children using available music production technology. The key component in the proposed prototype was the use of beat matched loops and effects which were synchronized to the songs tempo and made it possible for the children to express themselves musically.  The result also show that an important aspect of gaining well-being, motivation and meaningfulness among the children was connected to the proposed prototype ability to generate a musical outcome matching the children's expectations and personal preferences.

This Thesis deals with a method to enforce the intellectual property rights and protect digital media from tampering – Digital Audio Watermarking. The main aim of this work is implement an audio watermarking algorithm. The theoretical part defined basic terms, methods and processes, which are used in this area. The practical part shows a process of embedding the digital signature into a host signal and her backward extraction. The embedding rule used spread spectrum technique and a psychoacoustic model. The implemented psychoacoustic model involves two properties of the human auditory system which are frequency masking and representation the frequency scale on limited bands called critical bands. The model is relatively new and based on the DWPT. In terms of above model is then the digital watermark embedded in the wavelet domain. This algorithm is implemented in technical software MATLAB. One part of this work focuses on robustness tests of the algorithm. Common signal processing modifications are applied to the watermarked audio as follows: Cutting of the audio, re-sampling, lossy compression, filtering, equalization, modulation effects, noise addition. The last part of the thesis presents subjective and objective methods usable in order to judge the influence of watermarking embedding on the quality of audio tracks called transparency.

This thesis deals with design and development of a I2S module driver for real-time operating system MQX running on a ColdFire V4 architecture based processor. Further there are presented circuit diagrams and PCBs created for testing of functions and properties of the driver. Signal input is a digital audio in form of S/PDIF interface, output is an analogue signal, that can be listened by the means of a headphone amplifier. The conclusion describes a sample application demonstrating function of the driver on developed hardware.

The soundscape is a multi-sensory experience of an area. Previous research in the field has shown a positive correlation between designing with both audio and visual feedback for an increased perceived environmental pleasantness. With growing computer power, the possibilities of analysing audio increase the number of ways visual feedback can be adapted. However, little research has been done on the implementation of digital audio-visual computing for enhanced pleasantness of our sonic environment. This paper will therefore serve as a first step of investigating soundscapes from an audio-visual and digital perspective. A particle system was developed and tested in a user study in a static, moving and audio-reactive manner relative to a given soundscape. Quantitative analysis showed a decreased perception of negative sounds in an audio-reactive particle system, reactive to the positive sounds in the soundscape, compared to the static condition. Qualitative assessment indicated that different visual feedback shifts attention to specific sound sources, but the overall perception of the soundscape that leads to are varying. The findings in this study are useful for urban planners and designers who wish to explore soundscapes from a cross-disciplinary and multi-modal point of view. / Ljudlandskapet är en multimodal upplevelse av vår omgivning. Tidigare studier inom området har visat på en positiv korrelation mellan att designa med audiovisuella intryck och en förbättrad upplevelse av miljön omkring oss. Med konstant växande datorkraft ökar möjligheterna för att analysera ljud, och därmed möjligheterna att skapa visuellt material anpassat för den aktuella ljudmiljön. Däremot saknas forskning kring hur digitala audiovisuella medel kan implementeras för att förbättra upplevelsen av ett ljudlandskap. Den här artikel kommer därför fungera som ett första steg i att undersöka ljud i vår miljö från ett audiovisuellt och digitalt perspektiv. För att göra detta utvecklades ett partikelsystem som testades i ett statiskt, rörandes och audio reaktivt läge relativt till ett givet ljudlandskap. Kvantitativ analys visade på en minskad uppfattning av negativa ljud med ett partikelsystem audio reaktivt till de positiva ljuden i ett ljudlandskap, jämfört med ett statiskt. Kvalitativ bedömning indikerade att olika visuella intryck flyttar uppmärksamheten mot specifika ljudkällor, men den övergripande upplevelsen av ljudlandskapet är varierande. Resultaten av den här studien är användbar för stadsplanerare och designers som önskar att utforska ljudlandskap från ett tvärvetenskapligt och multimodalt perspektiv.