Loud stage sound is a known challenge for live sound engineers, and the goal of this study was to investigate stage sound's role in live rock music performed in small venues. Stage sound is known to be prominent in small rooms as the total sound energy produced tends to contain a large proportion of stage sound. It commonly addressed that if the PA is overpowered by direct sound from instruments on stage, the mixing engineer may lose control over musical balance and overall sound level. To understand how engineers handle this, 6 professional engineers were interviewed by using semi-structured interviews. Additionally, 14 shorter interviews were conducted with musicians to gain further insight in what attributes of stage sound are desirable for performers and how they are affected by engineers' decisions. Results were interpreted through thematic analysis. 8 main themes were identified, some of which contained several subthemes. These themes were 1) Managing instruments' stage sound, 2) Monitor systems, 3) PA systems - what sources need reinforcement?, 4) Strategies for even coverage, 5) Sound level management, 6) Limitations of small venues 7) What is "good sound"? and 8) Communication regarding stage sound levels.

One of the mixing engineers most important tool when listening, analyzing and taking decisions is the physical space in which the reproduction system and the monitors are placed. The space needs to fulfill certain aspects to be able to qualify as a control room considered good for taking these decisions. In this study an active listening test was conducted to investigate if the RT60 of the natural reverberation in the room affected a mixing engineers’ decisions when adjusting parameters on an artificial reverb. 16 subjects participated, the subjects adjusted reverb level and reverb time of an artificial reverb in two different acoustical environments, with two different values of RT60. The environments were based on a professional control room made for mixing and mastering. The two values for RT60 was achieved through manipulating the room with diffusors and absorbing material. It was found that the subjects were able to adapt to the different acoustical environments, although other differences were found. The order of the which of the environments the subjects started in and which parameter the subject started adjusting, showed a considerable impact on the result. As well as what kind of factors the subjects considered when adjusting the artificial reverb.

Visualization, processing and editing of audio, directly on a time-frequency surface, is the scope of this thesis. More precisely the scalogram produced by a Gabor Wavelet transform is used, which is a powerful alternative to traditional techinques where the wave form is the main visual aid and editting is performed by parametric filters. Reconstruction properties, scalogram design and enhancements as well audio manipulation algorithms are investigated for this audio representation.The scalogram is designed to allow a flexible choice of time-frequency ratio, while maintaining high quality reconstruction. For this mean, the Loglet is used, which is observed to be the most suitable filter choice.  Re-assignmentare tested, and a novel weighting function using partial derivatives of phase is proposed.  An audio interpolation procedure is developed and shown to perform well in listening tests.The feasibility to use the transform coefficients directly for various purposes is investigated. It is concluded that Pitch shifts are hard to describe in the framework while noise thresh holding works well. A downsampling scheme is suggested that saves on operations and memory consumption as well as it speeds up real world implementations significantly. Finally, a Scalogram 'compression' procedure is developed, allowing the caching of an approximate scalogram.

Monitoring is hugely important for musicians in any type of live situation for them to provide a good musical performance, and In-ear Monitoring (IEM) is increasingly popular among both musicians and audio engineers. Despite this, there has been little research on the topic and musicians and audio engineers would benefit from knowing more about the system, to utilize it in the best way possible. Because of the potentially added issues with structure borne sound, the study has chosen to focus on vocalists that use IEM. Through interviews with both Less experienced IEM users and Experienced IEM users, this study aims to find out what issues IEM users experience in live situations. Other than perceived issues, the study investigates how issues evolve with increased experience, comparing the two user groups, what solutions can be found and how an IEM mix should be for a vocalist. Through a Thematic analysis, the study reaches the conclusion that the main issue with the IEM system for vocalists is isolation from the surrounding world, and that the isolation leads to other perceived issues such as worsening the live feeling and obstructing contact with the audience. The study also concludes that there is a clear connection between monitoring and musical performance with poor monitoring leading to worse performance. The main solution observed is habit with the system, good communication between the audio engineers and musicians and an understanding for both sides workflow to increase efficiency in mixing. With this knowledge, further research can be made to increase the understanding for musicians and audio engineers respectively, and audio engineers and musicians can already focus on increasing cooperation.

Ce travail s’intéresse au problème de la rétroingénierie du son pour l’écoute active. Le format considéré correspond au CD audio. Le contenu musical est vu comme le résultat d’un enchaînement de la composition, l’enregistrement, le mixage et le mastering. L’inversion des deux dernières étapes constitue le fond du problème présent. Le signal audio est traité comme un mélange post-non-linéaire. Ainsi, le mélange est « décompressé » avant d'être « décomposé » en pistes audio. Le problème est abordé dans un contexte informé : l’inversion est accompagnée d'une information qui est spécifique à la production du contenu. De cette manière, la qualité de l’inversion est significativement améliorée. L’information est réduite de taille en se servant des méthodes de quantification, codage, et des faits sur la psychoacoustique. Les méthodes proposées s’appliquent en temps réel et montrent une complexité basse. Les résultats obtenus améliorent l’état de l’art et contribuent aux nouvelles connaissances. / 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.

Matched frequency responses are a fundamental starting point for a variety ofimplementations for microphone arrays. In this report, two methods for frequencyresponse-calibration of a pre-assembled microphone array are presented andevaluated. This is done by extracting the deviation in frequency responses of themicrophones in relation to a selected reference microphone, using a swept sine asa stimulus signal and an inverse filter. The swept sine includes all frequencieswithin the bandwidth of human speech. This allows for a full frequency responsemeasurements from all microphones using a single recording.Using the swept sine, the deviation in frequency response between the microphonescan be obtained. This deviation represents the scaling factor that all microphonesmust be calibrated with to match the reference microphone. Applying the scalingfactors on the recorded stimulus signal shows an improvement for both implementedmethods, and where one method matches the frequency response of the microphoneswith high accuracy.Once the scaling factors of the various microphones is obtained, it can be usedto calibrate other recorded signals. This leads to an minor improvement formatching the frequency responses, as it has been shown that the differencesin frequency response between the microphones is signal-dependent and variesbetween recordings. The response differences between the microphones dependson the design of the array, speaker, room and the acoustic frequency dispersionthat occurs with sound waves. This makes it difficult to calibrate the frequencyresponses of the microphones without appropriate equipment because the responseof the microphones is noticeably affected by these other factors. Proposals to addressthese problems are discussed in the report as future work. / Matchade frekvenssvar är en grundläggande utgångspunkt för ett flertal implementationer för mikrofonarrayer. I denna rapport presenteras och utvärderas tvåmetoder för frekvenssvarskalibrering för en förmonterad mikrofonarray. Detta görsgenom att extrahera avvikelsen i frekvenssvar hos alla mikrofoner i förhållandetill en vald referensmikrofon. Frekvenssvaren tas fram med hjälp av ettsinussvep som stimulanssignal och ett inverterat filter. Sinussvepet inkluderar helafrekvensbredden för mänskligt tal och möjliggör att mikrofonernas fulla frekvenssvarkan analyseras från en enda inspelning.Med hjälp av sinussvepet kan avvikelsen i frekvenssvar mellan mikrofonerna erhållas.Denna avvikelse representerar den skalningsfaktor alla mikrofoner måste kalibrerasefter för att matcha referensmikrofonen. Genom att applicera faktorerna på deninspelade stimulussignalen ses en förbättring för båda implementerade metoderna,där en metod matchar mikrofonernas frekvenssvar med hög noggrannhet.När skalningsfaktorn för de olika mikrofonerna har erhållits kan den användas föratt kalibrera andra inspelade signaler. Detta leder till en liten förbättring i att matchafrekvenssvaren, då det har visat sig att skillnader mellan mikrofonernas frekvenssvarär signalberoende och varierar mellan inspelningar. Skillnader i frekvenssvar mellanmikrofonerna beror på ljudets utbredning i rummet, utformningen av arrayen,högtalaren och den akustiska frekvensspridningen som uppstår hos ljudvågor. Dettagör det svårt att kalibrera frekvenssvaren hos mikrofonerna utan lämplig utrustningeftersom mikrofonernas respons märkbart påverkas av dessa andra faktorer. Förslagför att kringgå dessa problem diskuteras i rapporten och tas upp som framtidaarbete.