Denne oppgaven tar utgangspunkt i å undersøke hvordan ikke-ideelle lytterom påvirker lydsignalet i en auralisering, der auraliseringen blir presentert ved et konvensjonelt høyttalersystem. Metoden som blir tatt i bruk for å undersøke dette går ut på å kartlegge de viktigste akustiske parameterne ved en slik gjengivelse og hvor mye de endres som følge av å bli spilt av et høyttalersystem med Dolby 5.1 struktur. Parameterne det er fokusert på i denne oppgaven er: frekvensnivåer, ``Clarity'', etterklangstid og lateral fraction. For å undersøke rompåvirkningen i forskjellige tilfeller, er det valgt ut fem rom med noe ulik karakter, som alle er representative for realistiske rom å benytte til presentasjon av auralisering. Rommene gir fra helt tørr respons til responsen fra et auditorium og tre grader mellom. Syv simulerte rom av ulik karakter blir auralisert i Catt Acoustics og konvolvert med responsen fra sine respektive auraliseringskanaler i lytterommene. En omnidireksjonell mikrofon blir brukt for måling av frekvensrespons, C80 og T60. Og en Soundfield ambisonics mikrofon blir brukt til å måle LF. Konklusjonen i oppgaven er at de fleste av rommene ikke forringer den oktavmidlete frekvensgangen mer enn maksimalt 2dB i alle frekvensbånd, bortsett fra det laveste frekvensbåndet på 125Hz. T60 lar seg gjengi med rimelig nøyaktighet så lenge den auraliserte T60 verdien ligger godt over T60 verdien til lytterommet. De største avvikene ligger derimot i C80 verdiene og Lateral Fraction verdiene.

De senere årene har vist en stadig økende interesse for transistorer basert på GalliumNitrid, spesielt i design av effektforsterkere for trådløse applikasjoner. Denne rapportenbeskriver to klasse B forsterkerdesign for 2GHz basert på en 10W GaN-HEMT fra Cree. I tillegg presenteres relevant forsterkerteori, samt detaljerte beskrivelser av design-, konstruksjons- og måleprosessene.Det første forsterkerdesignet ble gjort ved bruk av en storsignalmodell fra Cree i Advanced Design System. Måleresultater viser god samsvar med simuleringsresultater for de to realiserte forsterkerne fra dette designet. Typiske avvik gjør seg likevel gjeldende, og denne oppgaven forsøker å kartlegge ulike faktorer i design-, konstruksjon- og måleprosessene som har betydelig innvirkning på de endelige resultatene.For å oppnå større korrelasjon, oppfordres det blant annet til å benytte alternative metoder ved produksjon av kretskort, verifisere nøyaktigheten til de passive komponentmodellene og kompensere for spenningsfall under effektmålinger. I 1dB-kompresjonoppnår en av de konstruerte forsterkerne fra dette designet en utgangseffekt på 39.8dBm, en forsterkning på 12.5dB, samt en effektivitet (PAE) på 55.2%.Det andre forsterkerdesignet ble gjort ved bruk av et moderne load-pull-oppsett, som gjør det mulig å utføre målinger på transistorterminalene i tidsdomenet. Tilpasningsnettverk på inn- og utgang ble dermed designet basert på måleresultater. Den realiserte forsterkeren fra disse målingene oppnår en noe lavere forsterkning, 9.7dB i 1dB-kompresjon, men med en høyere utgangseffekt på 41.2dBm og en noe bedre PAE på 56.7%. Med måleutstyret er det også gjort et studie av bølgeformene til strøm og spenning på drain som viser at denne forsterkeren er en harmonisk tunet klasse B forsterker.Rapporten viser med utgangspunkt i måleresultatene til den realiserte forsterkeren fra målingene at load-pull-oppsettet fremstår som et svært attraktivt alternativ til å benytte transistormodeller i Advanced Design System. Særlig dersom integrasjonen av måleutstyret i det eksisterende load-pull-oppsettet kan bedres og ved bruk av spenningsfallkompensering.

Denne oppgaven tar for seg teori, implementering, testing og mulighetene som ligger i målemetoden nærfelts akustisk holografi (NAH). Det har blitt konstruert et komplett målesystem med tanke på studie av en platehøyttaler. Teorien bak målemetoden vil bli presentert samt hvordan den implementeres i et fysisk apparat. Målemetoden har blitt testet ut både analytisk og praktisk og de endelige resultatene er meget positive. Hovedfokus for oppgaven har vært å studere metoden og undersøke noe av potensialet som ligger i den og det vil i den sammenheng pekes på endel videre arbeid som kan gjøres.

I dag er biofilm på proteser operert inn i kroppen årsak til infeksjonar og pasientsmerter. Ein ynskjer å vite om kavitasjon kan nyttast til å fjerne denne biofilmen. Tidlegare medisinske studiar med kommersielt tilgjengelege ultralydbad har ikkje klargjort om kavitasjonsfenomenet eksisterer i dei kolbene protesene må være i. Dersom kavitasjon kan nyttast kan dette gje ein raskare og rimelegare metode for å fjerne biofilm enn det som eksisterar i dag.For å vite om kavitasjon kan nyttast må ein vite om fenomenet oppstår i ein glaskolbe. Dette fordi protesene må liggje i sterile omgivnadar som ein slik kolbe er. Dette undersøkjast i eit konstruert ultralydbad der ein har kontroll på det akustiske lydtrykket og frekvensen. Eigenskapane til ultralydbadet samanliknast med kavitasjonsteori og eit kommersielt ultralydbad for å finne best mogleg oppsett av ultralydbadet og for å fastsetje eit eigna akustisk signal.Generelle førebuande målingar med fleire frekvensar blei gjort for å finne omtrentleg kavitasjonsterskel for så å kunne fortsette med fleire grundigare forsøk. Målingar blei gjennomført både med og utan glaskolbe for å sjå på skilnadane dette gav. Pulsbreidda til det akustiske signalet blei variert for å sjå effekten dette hadde på kavitasjonsterskel og for å sjå kor korte pulsar ein kan nytte og framleis få i gang kavitasjonsfenomenet. Dette for å kunne ha så lite kavitering som mogleg.Kavitasjon viste seg å vere mogleg inne i ein glaskolbe med akustisk trykk over unit{176}{kilopascal} for pulsar på 15 ms og ein reptisjonsrate på 10 Hz. Kavitasjon var også mogleg å få til for endå kortare pulsar med same repetisjonsrate, då med ein auka kavitasjonsterskel (opptil 240 kPa).Sidan resultata viser at vatnet kaviterer inne i ein glaskolbe er det grunn til å tru at dette fenomenet kan nyttast til å fjerne biofilm og at ein difor bør fortsette studiar på dette området.

The amount of non-textual media on the Internet is increasing, which creates a greater need of being able to search in this type of media. The goal with this thesis is to be able to do information search by use of soundtracks in audio databases. To get to know the content in an audio file, one wants a system that can automatically extract necessary information. The first step in making this system is to record what is happening at which time in an event log. This thesis treats the beginning of such a process. The experiments performed dealt with detection of pauses lasting longer than 1 second and detection of speaker changes. The corpus used in experiments consists of news broadcasts from The Norwegian Broadcasting Corporation (NRK) radio. Each broadcast had a transcription, which was used as a reference when evaluating the results. Another corpus, the HUB-4 1997 evaluation data, was used for comparative tests.A lot of work treating indexing of audio databases has already been conducted. As corpora are different, there may be varying results obtained from the same methods. In this thesis, common segmentation methods have been used with the parameters adapted to give as good results as possible with the given corpus. In the pause detection, model-based segmentation was used. A Gaussian mixture model was implemented for each of the two events: sound and long pause. For the speaker segmentation, experiments with different metric-based segmentation techniques were performed. The Bayesian information criterion (BIC) and a modified version of this criterion were tested with different options and parameter values. A false alarm compensation based on the symmetric Kullback-Leibler distance was implemented as an attempt to reduce the number of false change points. The pause detection was not successful. By using the manual transcription as reference, an F-score of 38.1 % was obtained when the settings were adjusted to result in about the same numbers for false alarms and false rejections. However, further investigation showed that the transcription had flaws with respect to labeling of pauses. An evaluation of the wrongly inserted pauses showed that most of these segments actually contained silence or noise. However, the number of pauses missed was unknown, and it was not possible to get a reliable F-score. An attempt on labeling all pauses in the HUB-4 1997 data was done. With the modified transcription, an F-score of 81.7 % was obtained. However, it is possible that unlabeled pauses still exist in the transcription, as the labeling was performed by only looking at the audio signal. From classification experiments it became clear that using 1st and 2nd order delta coefficients in the feature vectors gave an improvement over just using static MFCCs. An F-score of 98.8 % was obtained from these experiments, which implies that the models are good when the segment boundaries are known. In order to get trustworthy results from the recognition task, a review of the transcription must be done.When using the modified version of BIC and false alarm compensation for speaker change detection, an F-score of 77.1 % were obtained. The average mismatch between correctly detected change points and reference transcription was 339 milliseconds. As a measure of how good the algorithm is, an F-score of 72.8 % was obtained with the HUB-4 1997 data. Ajmera et al. (2002) obtained an F-score of 67 % with the same data. It became clear that full covariance matrices gave an improvement over diagonal covariance matrices and that static MFCCs as feature vectors gave better results than MFCCs including delta coefficients. Inclusion of pitch as another feature did not contribute to any improvement of the results.

Most automatic speech recognition systems are based on statistical models thatrequire training. While these types of systems have reached recognition ratesthat are sufficient for many purposes, they perform poorly for speaker typesthat are not present in the training material. Children are often absent fromtraining material for speech recognizers, and creating good training materialfor children can be difficult and expensive.To address this issue, this thesis focuses on using adult training material totrain a recognizer for children by adapting the training material duringtraining. Instead of performing speaker-dependent adaptation duringrecognition, where computational power may be scarce, and responsiveness may beessential, adaptation is performed during training towards a class of speakers.Using a combination of vocal tract length normalization (VTLN) and cepstralmean normalization during training, promising results have been obtained. In aconnected-digits task, a reduction in errors as high as 70% was shown, with areduction of almost 50% in a large vocabulary task. Using VTLN to warp thesame training material several times, combining these warped materials to trainone recognizer, a similar reduction in errors was shown, but with an increasedrobustness indicating a less speaker-dependent system. It is also shown that apiecewise linear warping method is better suited to warp adult speech to childspeech, than a bilinear warping method.

This report presents a potential method to efficiently implement edge diffraction from a noise barrier into geometrical room acoustic softwares. The modelling is based on semi-transparent surfaces, and the classic digital signal processing multipath transmission equation has been employed to describe in a mathematical term the presented method. The basic idea is to subdivide the noise barrier into a number of subareas, and then give each subarea an optimalized transmission coefficient for building the best possible output impulse response.To evaluate the proposed semi-transparent modelling, a Matlab simulation model of an infintite noise barrier case has been developed, and the corresponding simulations have been compared with the ideally correct solution. Accordingly, it is stated that there seems to be a clear positive potential in the proposed modelling technique. However, the results also reveal a somewhat instability in the modelling, which is expected to appear mainly for rare critical source and receiver positions.A main goal has been to develop a method that can easily be implemented in the existing calculation algorithms of today's commercial software developers. For verification, the proposed modelling has by discussion been associated with the often employed diffuse rain method. However, since no true implemenation in geometrical room acoustic software has been performed, further studies are required.To maintain efficiency and reliability, another desired outcome of the presented modelling has been that is should function for a general one-to-all source-receiver condition. Surely, the modelling seems to give fairly good results for symmetric source and receiver positions, but as the receiver is moved away from these symmetric conditions some unwanted errors occur, especially at higher frequencies.Main focus has been given to receiver positions located in the shadow zone, but some simulations and discussion has also been given to receiver positions located near the source-receiver sight line - at where direct sound energy contributions are also included and an interference pattern arises. To cope with this interference pattern, a polarity shift is proposed, which gives a clear improvement at low frequencies.One certainly interesting feature of the presented modelling technique is that it involves a broadband-based simulation method, which means that it gives the full frequency response by running only one simulation. Indeed, this is advantageous regarding calculation efficiency, but it does however also introduce some issues regarding a potential future software implementation - as the common case in geometrical room acoustics is to run individual octaveband-based simulations.

High frequency dispersion has a great influence on the perceived performance of a loudspeaker. The directivity of a single transducer primarily depends on driver size, however directivity can be modified using an acoustical waveguide. A method of modelling and designing a wide dispersion waveguide for a loudspeaker soft dome tweeter has been developed.A combination of finite element (FE) modelling and understanding of directivity and waveguides is used in order to prototype loudspeakers virtually. By utilizing computer simulations, the prototyping process is faster and more cost effective, all the while designing better performing loudspeakers.Firstly, a baseline acoustic-structure interaction FE model of a tweeter was built in the Comsol Multiphysics software. The model was verified by measurements, and the directional properties showed satisfactory agreement in the frequency range of interest. The accuracy of the baseline model allowed for credible simulations of waveguides.Secondly, many waveguide geometry types were investigated, and a method for randomizing geometries and automating the design process was developed using the Livelink for Matlab module in Comsol. Subsequently, a best fit waveguide design was selected based on a set of defined design criteria.Thirdly, a prototype was built, the measured performance compared to the simulated model, and discrepancies investigated. The waveguide directivity performs as modelled through most of the working range, although deviations from simulations were larger than expected at frequencies above 12 kHz. The measurements validate the modelling procedure and emphasize the value of the design algorithm, even though the prediction accuracy may be improved. It can be shown that a waveguide of this type can, with only small modifications, be an effective way to increase HF dispersion for a large range of commercially available tweeters.

An adaptable binaural model for surround sound has been developed in this master’s thesis. The adaptability is based on measurements of the listener’s head. This model is based on what was found to be the best suited material combination of successful models in earlier studies. This includes an ellipsoidal model for interaural time difference, an one-pole, one-zero head shadow filter and the use of Blauert’s directional bands for spectral manipulation. The model can play back six channel surround content using the standardized 5.1 surround sound loudspeaker setup. This standardized loudspeaker placement is used when creating virtual sound sources. Arbitrary sound directions are made in the horizontal plane by creating virtual sound sources using vector base amplitude panning between the standardized loudspeaker positions.To test the performance of this model, a listening test was conducted. The hypothesis tested was that the adaptable model would produce equal or lower localization error, compared to the commercial model. 20 test subjects participated. The test featured three different test types; standardized 5.1 loudspeaker setup, a commercial model for surround sound in headphones, and the adaptable model. Localization accuracy for ten selected directions in the right half plane was tested. The results from the adaptable model were compared to the result of the commercial model. The loudspeaker setup acted as a reference.Mean localization error was found to be thrice as high for the adaptable model, compared to the commercial model. Both models had the same standard deviation. 95% of the confidence intervals for these models did not overlap, i.e. there is a significant difference between the two methods. With this one can safely conclude that the commercial model provided a smaller localization error than the adaptable model. Hence the hypothesis has to be disproved.Both the commercial model and the thesis model performed significantly worse than the loudspeaker setup. One difference between commercial model, and the thesis model, was that that the commercial model had added room reflections and reverberation. This can create the sensation that the sound is coming from outside the head, and make it easier to localize. This contradicts with the knowledge that reverberation diffuses the sound field, making the direct sound that provides the directional information become less prominent.

This study has investigated to what extent musicians adjust their source levels to different music rehearsal rooms. In the experiment, six amateur musicians were to perform the same song i four different rehearsal rooms, by first singing, then by playing guitar and last by combining singing with guitar playing. All sound sources were recorded and analyzed. The results shows that the average musician adjusts his source levels to the rehearsal room and that most of the adjustments are made in the guitar playing. Looking at the individual musician there are some that do not show any signs as to being affected by the rooms, and there are some that shows clear signs of being affected by the rehearsal room. The result also shows that the musicians are affected differently by different acoustic parameters, whereas the strength shows the least correlation and reverberation time shows the most correlation to the adjustment made by the average musician.