The process of mixing tracks for a live stage performance or studio session is both time consuming and expensive with assistance of professionals. It is also difficult for individuals to remain competitive against established companies, since multiple tracks must be properly mixed in order to achieve well-enhanced elements -- generally, a poor mix makes it difficult for the listener to distinguish the different elements of the mix. The developed method during this thesis work aims at facilitating the mixing work for live performances and studio sessions. The implemented system analyzes the energy spectrum of the tracks included in the mix. By unmasking spectral components, the spectral overlap of the tracks is minimized. The system filters non-characteristic frequencies, leaving significant frequencies undisturbed. Five tracks have been used from the final mix of a successful radio song. These tracks have been analyzed and used to illustrate and validate the developed method. The system was successfully implemented in MATLAB with promising results and conclusions. The processed mix unmasks frequency content and is perceived to sound clearer compared to the unprocessed mix by a number of test individuals. The method reminds of a multi-band compressor that analyzes the spectral information between tracks. Thus, by use of inter-spectral dependencies, the thesis investigates the possibility to control the amplitudes in time by filtration in frequency domain. The compression rate in time domain is reflected in regard to a trade-off between conservation of characteristic frequencies and reduction of spectral overlaps.

<p>With the intention of surveying the field of research in adaptive audio systems for interactive media, a suggested audio system design for adaptive music control is described, and a prototype implementation of key parts of the system is presented and evaluated. </p><p>Foregoing midi-triggered sound banks, the proposed design uses layered segmented audio files, defined and controlled by XML-scripts. The results demonstrate an inclination of a flexible system, capable of adequate adaptive behaviour of high quality sound. </p><p>The implemented system will serve as an extensive basis for future work contributing to the research of adaptive behaviour to both music and sound effects for interactive media, and also as a preliminary foray into the more experimental field of stand-alone non-linear music playback.</p>

With the intention of surveying the field of research in adaptive audio systems for interactive media, a suggested audio system design for adaptive music control is described, and a prototype implementation of key parts of the system is presented and evaluated. Foregoing midi-triggered sound banks, the proposed design uses layered segmented audio files, defined and controlled by XML-scripts. The results demonstrate an inclination of a flexible system, capable of adequate adaptive behaviour of high quality sound. The implemented system will serve as an extensive basis for future work contributing to the research of adaptive behaviour to both music and sound effects for interactive media, and also as a preliminary foray into the more experimental field of stand-alone non-linear music playback.

The focus of this thesis is to study the way that adaptive audio can be used in digital games and how it can be used to portray different moods to the listener, how games can reflect different feelings and how quickly those feelings can change. Games audio environment is significant to be able to adapt to the ever changing narrative of the game. The purpose is to gain insight in how immersion in digital games can be improved with the use of adaptive audio ­ to study if there’s an easy way to implement a system where audio can be mixed and adjusted in real­time to mirror the events in the game and project accurate feelings. To study this we will create a parameter based system in the sound engine of the game we will make during the production phase, with different parameters based on a number of different factors . Keywords: Adaptive Audio, Parameter System, Mood Music, Digital Games, Game Production Syftet med detta kandidatarbete är att studera hur adaptiva ljud kan användas i digitala spel, hur de kan användas för att beskriva olika stämningar till lyssnaren, och hur spel kan spegla olika känslor och hur snabbt dessa känslor kan förändras. Det är viktigt att ett spels ljudmiljö kan anpassa sig till den ständigt föränderliga berättelsen i spelet. Syftet är att få en inblick i hur immersion i digitala spel kan förbättras med användning av adaptiva ljud ­ att studera om det finns ett enkelt sätt att implementera ett system där ljudet kan mixas och anpassas i realtid för att spegla händelserna i spelet och återskapa äkta känslor. För att undersöka detta kommer vi skapa ett parameterbaserat system i ljudmotorn till spelet som vi kommer göra under produktionsfasen, med olika parameterar som är baserade på ett antal olika faktorer. Nyckelord: Adaptivt Ljud, Parametersystem, Stämningsmusik, Digitala Spel, Spelproduktion. / Ett arbete som tar upp hur man kan få immersionen i digitala spel att förbättras med användning av adaptivt ljud och musik för att spegla narrativet. / Pontus Stensgaard: 0769123182 pontus.stensgaard@gmail.com

Note sequence morphing is the combination of two note sequences to create a ‘hybrid transition’, or ‘morph’. The morph is a ‘hybrid’ in the sense that it exhibits properties of both sequences. The morph is also a ‘transition’, in that it can segue between them. An automated and interactive approach allows manipulation in realtime by users who may control the relative influence of source or target and the transition length. The techniques that were developed through this research were designed particularly for popular genres of predominantly instrumental electronic music which I will refer to collectively as Mainstream Electronic Music (MEM). The research has potential for application within contexts such as computer games, multimedia, live electronic music, interactive installations and accessible music or “music therapy”. Musical themes in computer games and multimedia can morph adaptively in response to parameters in realtime. Morphing can be used by electronic music producers as an alternative to mixing in live performance. Interactive installations and accessible music devices can utilise morphing algorithms to enable expressive control over the music through simple interface components. I have developed a software application called LEMorpheus which consists of software infrastructure for morphing and three alternative note sequence morphing algorithms: parametric morphing, probabilistic morphing and evolutionary morphing. Parametric morphing involves converting the source and target into continuous envelopes, interpolation, and converting the interpolated envelopes back into note sequences. Probabilistic morphing involves converting the source and target into probability matrices and seeding them on recent output to generate the next note. Evolutionary morphing involves iteratively mutating the source into multiple possible candidates and selecting those which are judged as more similar to the target, until the target is reached. I formally evaluated the probabilistic morphing algorithm by extracting qualitative feedback from participants in a live electronic music situation, benchmarked against a live, professional DJ. The probabilistic algorithm was competitive, being favoured particularly for long morphs. The evolutionary morphing algorithm was formally evaluated using an online questionnaire, benchmarked against a human composer/producer. For particular samples, the morphing algorithm was competitive and occasionally seen as innovative; however, the morphs created by the human composer typically received more positive feedback, due to coherent, large scale structural changes, as opposed to the forced continuity of the morphing software.