The present thesis establishes a complete framework for the combination of finite element and classical ray based acoustic simulations in small rooms and discusses the inherent challenges and limitations including all aspects of sound generation, sound reflection and sound reception. In this context, the thesis gives detailed guidelines for the best-possible determination of all necessary input data for both simulation domains. The overall potential of the presented combined approach is assessed by conducting extensive objective and subjective comparisons of measurement and simulation results for three types of acoustically relevant small spaces (a scale-model reverberation room, a recording studio and two different car passenger compartments).

Starting from physical theory, this work develops a novel framework for the acoustic simulation of sound radiation by loudspeakers and sound reinforcement systems. First, a theoretical foundation is derived for the accurate description of simple and multi-way loudspeakers using an advanced point-source ''CDPS'' model that incorporates phase data. The model's practical implementation is presented including measurement requirements and the GLL loudspeaker data format specification. In the second part, larger systems are analyzed such as line arrays where the receiver may be located in the near field of the source. It is shown that any extended line source can be modeled accurately after decomposition into smaller CDPS elements. The influence of production variation among elements of an array is investigated and shown to be small. The last part of this work deals with the consequences of fluctuating environmental conditions such as wind and temperature on the coherence of sound signals from multiple sources. A new theoretical model is developed that allows predicting the smooth transition from amplitude to power summation as a function of the statistical properties of the environmental parameters. A part of this work was distinguished with the AES Publications Award 2010. Parts of the proposed data format have been incorporated into the international AES56 standard.

Measured transfer functions of acoustic systems are often used to derive single-number parameters. The uncertainty analysis is commonly focused on the derived parameters but not on the transfer function as the primary quantity. This thesis presents an approach to assess the uncertainty contributions in these transfer functions by using analytic models. Uncertainties caused by the measurement method are analyzed with a focus on the underlying signal processing. In particular, the influence of nonlinearities in the acoustic measurement chain are modeled to predict artifacts in the measured signals and hence the calculated acoustic transfer function. Secondly, characterization methods commonly applied in the field of signal processing are linked to the acoustic scenarios and the main influencing parameters. Acoustic parameters are then derived analytically and by means of Monte Carlo simulations considering the uncertainty of these input parameters. In order to provide airborne applications, analytic models for sound barrier and room acoustic measurements are developed incorporating the directivity and the orientation of the sound source as well as the positions of sources and receivers. The simulated uncertainty contributions are validated by measurements. The same approach is also applied to structure-borne sound applications.

Human sound localization helps to pay attention to spatially separated speakers using interaural level and time differences as well as angle-dependent monaural spectral cues. In a monophonic teleconference, for instance, it is much more difficult to distinguish between different speakers due to missing binaural cues. Spatial positioning of the speakers by means of binaural reproduction methods using head-related transfer functions (HRTFs) enhances speech comprehension. These HRTFs are influenced by the torso, head and ear geometry as they describe the propagation path of the sound from a source to the ear canal entrance. Through this geometry-dependency, the HRTF is directional and subject-dependent. To enable a sufficient reproduction, individual HRTFs should be used. However, it is tremendously difficult to measure these HRTFs. For this reason this thesis proposes approaches to adapt the HRTFs applying individual anthropometric dimensions of a user. Since localization at low frequencies is mainly influenced by the interaural time difference, two models to adapt this difference are developed and compared with existing models. Furthermore, two approaches to adapt the spectral cues at higher frequencies are studied, improved and compared. Although the localization performance with individualized HRTFs is slightly worse than with individual HRTFs, it is nevertheless still better than with non-individual HRTFs, taking into account the measurement effort.

In this thesis, acoustic measurement techniques related to the absorbing as well as the scattering properties of architectural surfaces are investigated. The research is divided into two parts: the first part consists of determining the most relevant causes of uncertainty for the standardized measurement methods of random-incidence absorption and scattering coefficients. A method is developed to determine the necessary minimum number of source-receiver combinations in the reverberation chamber to ensure a specified precision of the absorption or scattering coefficient. The second part of the thesis focuses on a hemispherical microphone array and signal processing steps related to the measurement of angle-dependent reflection properties. Measurements show that the array setup can be used to obtain the angle-dependent absorbing properties of samples with few source positions. With the help of array signal processing methods, the setup can also be used to determine the directional diffusion and scattering coefficient of small samples, yielding the same result as established far-field methods.

Structure-borne sound sources are vibrational sources connected in some way to the building structure. The mechanical excitation of the building structure leads to sound radiation. This is an important source of annoyance in modern light-weight buildings. The prediction of the sound pressure level from structure-borne sound sources is highly complicated because of the complexity involved in the coupling between source and receiver structure. The current standard on characterisation of service equipment in buildings EN 12354-5, can deal with sources on heavy structures (high-mobility source) but to date, there is no engineering method available for the case of coupling between source and receiver. A case study of a washing machine on a wooden joist floor is investigated in this thesis. In the first part, measurements in the coupled state are conducted. It is shown that the normal components are sufficient to predict the sound pressure level. However, this only applies to the coupled state. In the second part, a true prediction is calculated from independently measured source and receiver quantities. The difference between predicted and directly measured sound pressure level leads to considerable errors of up to 20 dB at low frequencies. This shows that the normal components are not sufficient to predict the coupling between a washing machine and a wooden floor.

In this work the importance of individualization in binaural technique is investigated. The results extend the present knowledge on the efficient measurement of individual head-related transfer functions (HRTFs) and highlight the importance of individual equalization filters in binaural reproduction, using both loudspeakers and headphones. Moreover, an integrated framework for the calculation of such equalization filters is presented. An innovative measurement setup was developed to allow the fast acquisition of individual HRTFs. The hardware was designed to be compatible with the range extrapolation technique. An individual HRTF dataset with 4000 directions can be measured in less than 6 minutes with this new setup. A framework was presented that integrates causality constraints to the regularized frequency domain calculation of crosstalk cancellation (CTC) filter. This framework also addresses the switching of active loudspeakers applying a weighted filter calculation method. A sound localization test showed that individualized CTC systems provide performance similar to that of binaural listening while nonindividualized CTC systems provide a significantly lower localization performance. Finally, a robust individual headphone equalization method was proposed. Perceptual tests showed that, in all but one of the tested situations, no audible differences between the original sound source and its binaural auditory display could be perceived.

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In this work, the possibilities of an acoustic field analysis in small microphone arrays are investigated. With the increased use of mobile communication devices, such as smartphones and hearing aids, and the increase in the number of microphones in such devices, multi-channel signal processing has gained popularity. Apart from the definite signal processing, this thesis evaluates what information on the acoustic sound field and environment can be gained from the signal of such small microphone arrays. For this purpose, an innovative sound field classification was developed that determines the energies of the single sound field components. The method is based on spatial coherences of two or more acoustical. The method was successfully verified with a set of simulated and measured input signals. An adaptive automatic sensor mismatch compensation was created, which proved able to fully compensate any slow sensor drift after an initial training. Further, a new method for the blind estimation of the reverberation time based on the dependency of the coherence estimate on the evaluation parameters was proposed. The method determines the reverberation time of a room from the spatial coherence between two or more acoustic sensors.