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An international standard concerned with the calibration of microphones in a free field by comparison has recently been published. The standard contemplates two main calibration methodologies for determining the sensitivity of a microphone under test when compared against a reference microphone. The two methodologies assume that the two microphones are exposed to the same sound pressure. This can be achieved by measuring the ratio of output voltages either sequentially or simultaneously. The first method requires a stable source to ensure that the sound pressure is approximately the same when the reference and test microphones are measured, whereas the second requires a source with a symmetrical directivity that ensures that the microphones placed at opposite positions are subjected to the same sound pressure. The two methods have been investigated experimentally in an extended frequency range. A third method, consisting of a combination of the sequential and simultaneous methodologies, has also been investigated. Though the application of time selective techniques is not discussed, the experimental results indicate the immunity to unwanted reflections in the sequential and combined approaches while it may be necessary to apply these techniques in the simultaneous approach.

Beamforming techniques are usually based on microphone arrays. The present work uses a beam of light as a sensor element, and describes a beamforming system that locates sound sources basedon the acousto-optic effect, this is, the interaction between sound and light. The use of light as a sensing element makes this method immune to spatial aliasing. This feature is illustrated by means ofsimulation and experimental results. For ease of comparison, the study is supplemented with results obtained with a line array of microphones

This investigation is concerned with generating a controlled sound field for listeners inside a circular array of loudspeakers without disturbing people outside the array. Ideally this configuration would have the advantage that reflections from the surroundings would be of no concern. Inspired by the Kirchhoff-Helmholtz integral theorem a double-layer array of loudspeakers is used. Several solution methods are suggested and examined with computer simulations: pure contrast control, pure pressure matching, and a weighted combination. In order to compare the performance of the methods two performance indices are used, i) the ratio of the sound energy in the listening zone to the sound energy in the quiet zone, and ii) a normalised measure of the deviations between the desired and the generated sound field in the listening zone. The best compromise is obtained with the method that combines pure contrast control with a pressure matching technique.

The influence of turbulent boundary layer pressure fluctuation on the sound intensity measurement in a flow is a subject of practical concern, because the sound intensity probe is generally exposed to the airflow and is sensed the turbulent boundary layer (TBL) pressure fluctuation which may even overwhelm the true source pressure in some cases. In this paper, the model of the sound intensity caused by the TBL pressure fluctuation is described firstly. Based upon the developed model, the sound intensity caused by the TBL pressure fluctuation is calculated using the available models of the wave-vector frequency spectra of the TBL pressure fluctuation. In order to validate the model and the numerical results, a serious of measurements were carried out. It is shown that the calculated results of the TBL pressure fluctuation agree fairly well with the measured results which are corrected with the estimated spatial response function of the microphone. Also, the characteristics of the measured sound intensity are consistent with that of the calculated sound intensity.

Many acoustic measurements rely on determining the total sound energy in an enclosure; and this quantity is usually estimated by measuring the mean square pressure at a number of discrete positions. Almost 30 years ago it was shown theoretically that the normalised spatial variance of the total sound energy density (potential and kinetic) is one third of the normalised spatial variance of the potential energy density (the mean square pressure) in a reverberant sound field above the Schroeder frequency. About ten years later this prediction was confirmed experimentally. However, until recently measurement of the total sound energy density (in air) has required an elaborate arrangement based on finite different approximations using at least four matched pressure microphones; therefore the method has never come into use. However, with the advent of a three-dimensional particle velocity transducer it has become somewhat easier to measure total rather than only potential energy density in a sound field. This paper examines the spatial uniformity of potential, kinetic and total sound energy density in enclosures theoretically and experimentally with particular emphasis on the frequency range below the Schroeder frequency.

Spherical near field acoustic holography (SNAH) is a recently developed technique that makes it possible to reconstruct the sound field inside and just outside an acoustically transparent spherical surface on which the sound pressure is measured with an array of microphones with negligible scattering. Because of the versatile geometry of a sphere SNAH is potentially extremely useful for source identification. On the other hand a rigid sphere is somewhat more practical than an open sphere, and it is possible to modify the SNAH theory so that a similar sound field reconstruction can be made with an array of microphones flush-mounted on a rigid sphere. However, this approach is only valid if it can be assumed that the sphere has a negligible influence on the incident sound field, in other words if multiple scattering can be ignored, and this is not necessarily a good assumption when the sphere is close to a radiating surface. This paper describes the modified SNAH theory and examines the matter through simulations and experimentally.

In buildings built with new techniques there exists a need for better understanding of their acoustical performance. The development of large wooden houses slows down by the uncertainty and costly testing that have to be performed many times before gaining good results. A greater understanding of the sound insulation for lightweight buildings have the possibility to speed up the development of new techniques and in the end give tenants better quality of life. This study uses Fourier sinus series to calculate the vibrations on a rib stiffened plate. The beams are modelled as line forces and moments that reacts onto the plate vibrations. A parameter study is performed with the aim to identify the most important parameters and their behaviour. The preliminary results show that the attenuation of the system is by far most evident in the direction across the beams. The influence from the basic input parameters on the attenuation is then studied. And it is preliminary shown that the placement of the excitation force within a bay actually is irrelevant for the calculated attenuation.

For some time now, statistical analysis has been a valuable tool in analyzing room transfer functions (RTFs). This work examines existing statistical time-frequency models and techniques for RTF analysis (e.g., Schroeder's stochastic model and the standard deviation over frequency bands for the RTF magnitude and phase). RTF fractional octave smoothing, as with 1-slash 3 octave analysis, may lead to RTF simplifications that can be useful for several audio applications, like room compensation, room modeling, auralisation purposes. The aim of this work is to identify the relationship of optimal response smoothing (e.g., as in complex smoothing) with respect to the original RTF statistics. More specifically, the RTF statistics, derived after the complex smoothing calculation, are compared to the original statistics across space inside typical rooms, by varying the source, the receiver position and the corresponding ratio of the direct and reverberant signal. In addition, this work examines the statistical quantities for speech and audio signals prior to their reproduction within rooms and when recorded in rooms. Histograms and other statistical distributions are used to compare RTF minima of typical “anechoic” and “reverberant” audio speech signals, in order to model the alterations due to room acoustics. The above results are obtained from both in-situ room response measurements and controlled acoustical response simulations.

Several variants of near field acoustic holography (NAH) that do not require measurement areaslarger than the source have been proposed. This paper examines and compares two such methods,statistically optimised near field acoustic holography (SONAH) and the Helmholtz equation leastsquares method (HELS). The HELS method is based on expanding the sound field into spherical harmonicsand may therefore be expected to work best for spherical sources; SONAH is based on propagatingand evanescent plane waves and may be expected to work best for planar sources. The performanceof SONAH and HELS is examined for spherical as well as non-spherical sources.

It is well know that the weakest link in a sound reproduction chain is the loudspeaker. The most significant effect on the sound quality is nonlinear distortion of loudspeakers. Many methods are applied to analyze the nonlinear distortion of loudspeakers. Almost all of the methods are based on the Fourier transform. In this work, a new method using the instantaneous frequency is introduced for describing and characterizing loudspeaker nonlinearities. First, numerical integration is applied to simulate the nonlinearities of loudspeakers caused by two nonlinear parameters, force factor and stiffness, separately. Then, a practical loudspeaker is used in an experiment and its nonlinear characteristics are analyzed with the instantaneous frequency. The results provide a clear physical interpretation of the nonlinearities of loudspeakers and will be useful for understanding the nonlinear behavior of loudspeakers. It will also be helpful for compensating for the nonlinearities and for improving the quality of sound reproduction. [Work supported by Sino-Danish International Co-operative Project, No. AM13: 66 and DANIDA (Denmark).]

Numerical methods based on the Helmholtz integral equation are well suited for solving acoustic scattering and diffraction problems at relatively low frequencies. However, it is well known that the standard method becomes degenerate if the objects that disturb the sound field are very thin. This paper makes use of a standard axisymmetric Helmholtz integral equation formulation and its boundary element method (BEM) implementation to study the behavior of the method on two test cases: a thin rigid disk of variable thickness and two rigid cylinders separated by a gap of variable width. Both problems give rise to the same kind of degeneracy in the method, and modified formulations have been proposed to overcome this difficulty. However, such techniques are better suited for the so-called thin-body problem than for the reciprocal narrow-gap problem, and only the first is usually dealt with in the literature. A simple integration technique that can extend the range of thicknesses/widths tractable by the otherwise unmodified standard formulation is presented and tested. This technique is valid for both cases. The modeling of acoustic transducers Like sound intensity probes and condenser microphones has motivated this work, although the proposed technique has a wider range of applications.

Plane wave propagation in ducts with rigid walls, radiation from ducts, classical four-pole theory for composite duct systems, and three-dimentional waves in wave guides of various cross-sectional shape are described.