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.

A video conferencing situation combines the acoustical properties of two rooms. As the talker is located in a one room, the sound reaching the listener in an other room will be colored by the acoustical characteristics of both of them. This thesis aims to survey the current conditions in a selection of video conferencing rooms, by investigating several room acoustic parameters involving background noise, re- verberation time, speech clarity and speech intelligibility. Convolution of recorded impulse responses enables the rooms to be combined, and the combined results to be evaluated. The evaluation of the results allow limit values for acceptable quality for video conferencing to be suggested. Limits for the highest acceptable values for the early decay time and the lowest acceptable values for speech clarity are sug- gested both for the single-room situation, and for the combination of rooms. The suggested values are based on specifications from building standards and relations between measured room acoustic descriptors.

The objective of this assignment was to implement a bistatic radio acoustic sounding system (RASS) in a anechoic chamber, using equipment at hand. The purpose of which was to enable measurements within a controlled environment to avoid variations in wind, humidity and temperature. Radio acoustic sounding systems as well as planning of the test setups have been theoretically investigated in previous work.The approach to the task has been to choose equipment assumed suitable for the experiment and verify the equipment suitability by measuring the technical characteristics relevant to the task. Then a bistatic RASS was constructed within a anechoic chamber. Using software defined radios, the received signals were digitally sampled. So was the acoustic field in the volume of interaction.Although processing the results failed to detect any sign of scatter, the equipment has been thoroughly tested. The acoustic, as well as electromagnetic equipment have shown good results with respect to the suitability for such a system prototype.No conclusion as to whether any scattering occurs as a consequence of the acoustic field can be made, as the absence of such indications may merely indicate insufficient sending power or too high levels of background noise. Prior to these findings, the equipment were concluded to be suitable for a prototype bistatic RASS. Methods to improve the measurement quality have been suggested as future work.

In this report various techniques used to estimate head related impulse responses are compared. The purpose is to investigate the effectiveness of presenting auralization via the QuietPro system's earplugs, and see if sound localization in the horizontal plane is possible. In addition, a theoretical study to relate pinna dimensions to features found in measured head related impulse responses is described. In the theoretical part, impulse responses for 33 left ears found in the CIPIC database were investigated, as an attempt to relate reflection coefficients and time delays associated with reflections from the pinna to physical dimensions of the ear. Unfortunately, no clear connection was found.In the listening test, the participants were sitting in the middle of a circle, surrounded by 36 numbered pieces of paper (either standing on top of e.g loudspeakers or attached to microphone stands) that indicated possible sound directions. 14 subjects performed sound localization tests by listening to three consecutive noise bursts of 150 ms duration with 100 ms silence between. Prior to the experiment, measurements of the subject's head were made and used for customization of the models. The task was to determine which of the 36 possible directions the sound was meant to come from. Seven simulation conditions were evaluated, each including 33 stimuli. Four test stimuli were also presented, resulting in a total of 235 noise bursts for each subject.The results show that the presented methods provide directionality to the stimuli, and that sound localization is possible. However, a significant reduction in localization performance compared to what could be expected for normal hearing conditions is observed. A high number of front/back confusion is reported, and even some instances of left/rigth confusion. Accuracy of the results was not predicted by model complexity, and in some cases it turned out that adding more features significantly degraded the performance.

SammendragDenne oppgaven ser på metoder for databasert prediksjon av spredningsfaktorer. Teorien bak dette har blitt presentert. Det har blitt utarbeidet en metode for å lage måleserier med impulsresponser langs en linje i et rom. Disse måleseriene har blitt lastet inn i Matlab. Her har det har blitt utarbeidet en metode for etterbehandling av disse, med den hensikt å hente ut hver enkelt bølgefront fra den samlede matrisen med alle impulsresponsene. Dette har gjort det mulig å studere hver enkelt bølgefront i detalj og sammenligne de med simulerte verdier. Feilkilder i forbindelse med målingene og etterbehandlingen har også blitt vurdert. Det pekes til slutt på en del videre arbeid som kan være aktuellt å gjøre i framtiden for å utvikle og studere metoden enda nærmere.

Today's audio in video-conference rooms do not in general have high quality audio standards.Most of the set-ups are PC-Speakers mounted on the wall, with a microphone on the table. With this, strong room modes are often excited from the speakers. When speaking to someone which also have bad equipment, the speech-intelligibility can be really bad. One solution presented with this Master thesis is to improve the loudspeaker set-up by mounting a DML(Distributed Mode Loudspeaker) above and parallel to the conference table. The DML will then radiate sound on both sides of the table equally in opposite phase, like a dipole. This will also minimize sound-radiation to the ceiling and the office-table.The DML is designed and modified from a loudspeaker originally deigned by the loudspeaker-company e-Scape. The Plexiglas panel, made out of an Acrylic material, has exciters mounted on the panel. After testing different combinations of two and three exciters in an anechoic chamber, the polar frequency response of the DML with only one exciter had a much better response than the other combinations.The idea of mounting the DML over the table is that the audience will get closer to the sound source, and room-effects will be very small compared to the direct signal. This should give a much better speech-intelligibility of the perceived sound signal compared to existing solutions. Subjective tests show that the majority of the participants preferred the DML rather than the PC-Speaker in all areas; a more natural sound, more closeness to the one speaking and better speech-intelligibility.One problem, which is seen on the measurements and feedback from the participants on the subjective test, is a low SPL(Sound Pressure Level) in the low frequency area. Under 100Hz the magnitude of the DML is reduced, compared to the response of ordinary monopole speakers. From this frequency area one can also see some dipole effects from the results, which gives us good qualities such as less radiation to the ceiling and the office table.

I små rom, kan fenomenet resonans være årsak til ugunstige akustiske forholdved at visse frekvenser blir overdrevent kraftige mens andre toner blir knapthørbare. Spesielt gjelder dette rom tiltenkt akustisk bruk, som opptaksstudioerog kontrollrom. Problemet kan unngås ved akustisk behandling i ettertid og/ellerarkitektonisk planlegging, og begge deler kan dra nytte av datasimulering. Tilgjengeligesimuleringsverktøy er konsentrert på mellom- og høyfrekvensområdet, ogdekker ikke lavfrekvente bølgers diffraktive oppførsel. Akselerasjon av parallelleberegninger på GPU tillater derimot hurtig og presis simulering, med metodeneFDTD og FDFD. En programvare beskrives og delvis utvikles i C#. Den drarnytte av GPU-en gjennom Cudafy via CUDA. Den vil forventes å kunne assistereved plassering av høyttaler, lyttepunkt og bassabsorbenter, såvel som i geometriskutforming av et rom.I denne oppgaven betraktes resonansproblemet først fra et erfaringsperspektiv,liknende situasjoner og mangelen på reelle løsninger identifiseres, og dette dannerbakgrunnen for arbeidet. Videre behandles fenomenet resonans teoretisk, i enkletermer og med flere eksempler. Dette etterfølges av en matematisk-teoretiskgjennomgang av simuleringsmetodene FDTD og FDFD. Til slutt beskrivesimplementasjonsdetaljer og utarbeidede løsninger, samt de uløste utfordringene– især FDFD-implementasjon – som gjenstår før programmet kan bli en realitet.

<p>Blind spectrum sensing in cognitive radio is being addressed in this thesis. Particular emphasis is put on performance in the low signal to noise range. It is shown how methods relying on traditional sample based estimation methods, such as the energy detector and autocorrelation based detectors, suffer at low SNRs. This problem is attempted to be solved by investigating how higher order statistics and information theoretic distance measures can be applied to do spectrum sensing. Results from a thorough literature survey indicate that the information theoretic distance gls{kl} divergence is promising when trying to devise a novel cognitive radio spectrum sensing scheme. Two novel detection algorithms based on Kullback-Leibler divergence estimation are proposed. However, unfortunately only one of them has a fully proven theoretical foundation. The other has a partial theoretical framework, supported by empirical results. Detection performance of the two proposed detectors in comparison with two reference detectors is assessed. The two reference detectors are the energy detector, and an autocorrelation based detector. Through simulations, it is shown that the proposed KL divergence based algorithms perform worse than the energy detector for all the considered scenarios, while one of them performs better than the autocorrelation based detector for certain signals. The reason why the detectors perform worse than the energy detector, despite the good properties of the estimators at low signal to noise ratios, is that the KL divergence between signal and noise is small. The low divergence stems from the fact that both signal and noise have very similar probability density distributions. Detection performance is also assessed by applying the detectors to raw data of a downconverted UMTS signal. It is shown that the noise distribution deviates from the standard assumption (circularly symmetric complex white Gaussian). Due to this deviation, the autocorrelation based reference detector and the two proposed Kullback-Leibler divergence based detectors are challenged. These detectors rely heavily on the aforementioned assumption, and fail to function properly when applied to signals with deviating characteristics.</p>

<p>Computer Assisted Pronunciation Training systems have become popular tools to train on second languages. Many second language learners prefer to train on pronunciation in a stress free environment with no other listeners. There exists no such tool for training on pronunciation of the Norwegian language. Pronunciation exercises in training systems should be directed at important properties in the language which the second language learners are not familiar with. In Norwegian two acoustically similar words can be contrasted by the vowel length, these words are called vowel length words. The vowel length is not important in many other languages. This master thesis has examined how to make the part of a Computer Assisted Pronunciation Training system which can evaluate non-native vowel length pronunciations. To evaluate vowel length pronunciations a vowel length classifier was developed. The approach was to segment utterances using automatic methods (Dynamic Time Warping and Hidden Markov Models). The segmented utterances were used to extract several classification features. A linear classifier was used to discriminate between short and long vowel length pronunciations. The classifier was trained by the Fisher Linear Discriminant principle. A database of Norwegian words of minimal pairs with respect to vowel length was recorded. Recordings from native Norwegians were used for training the classifier. Recordings from non-natives (Chinese and Iranians) were used for testing, resulting in an error rate of 6.7%. Further, confidence measures were used to improve the error rate to 3.4% by discarding 8.3% of the utterances. It could be argued that more than half of the discarded utterances were correctly discarded because of errors in the pronunciation. A CAPT demo, which was developed in an former assignment, was improved to use classifiers trained with the described approach.</p>