Technical Reviews

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Heat stress is a major problem in several working environments. Although technology has brought about remote control of industrial processes from airconditioned cabins, there are still many people who have to work in hot environments. Several methods for evaluation of heat stress have been presented In the literature.

This article describes the method of evaluation of heat stress by means of the WBGT -index, which has been standardized recently in the International Standard ISO 7243. Physiological reactions in hot environments are shortly described. Through the years several heat stress indices (both empirical and analytical) have been developed and used. The most common are also presented here.

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This article investigates the capability of an array containing two parallel layers with 8 × 8 microphones in each layer and with 3 cm microphone pitch to measure sound intensity at low frequencies in sound fields with strong reactive or diffuse components. This is done basically by extending the concept of pressure-residualintensity (p-RI) index from two-microphone intensity probe measurements to array measurements and by estimation of that index based on both simulated and real measurements. Good agreement is achieved between p-RI indices obtained from simulated measurements and from real measurements in a large standing wave tube. It is shown that for typical array microphones it is essential to correct for the differing frequency responses, typically based on response data stored in the Transducer Electronic Data Sheets (TEDS) of the individual microphones.

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Time Selective Response, TSR, is a frequency response measurement method based on linearly swept sine signals. Because TSR can be used for free-field measurements in ordinary rooms and is fast, accurate and relatively insensitive to background noise, it is convenient for measurements of microphone and sound level meter free-field responses. However, the method’s combination of time and frequency weighting can make it complicated to estimate the uncertainty of the measured response. This paper briefly summarizes the method and presents some experience with and guidelines for choosing measurement and weighting parameters and considerations on the associated uncertainty on the results. The results are discussed on the basis of practical measurements of microphone and sound level meter free-field responses at Brüel&Kjær.

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This paper describes the design of a Volume Velocity Source and verifies its use in the measurement of p/Q FRFs. Two designs of Volume Velocity Source are presented, one for low-mid frequencies, and one for mid-high frequencies, both taking advantage of the two-microphone technique for estimating volume velocity at the orifice during operation.

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Laser interferometry – using counters for fringe counting and determination of zeroes at higher frequencies – has been used for accelerometer calibration since the late 1960s. As digital techniques evolved during the 1980s and 90s, it became possible to make a more sophisticated approach towards interferometer detectors’ complex output signals. This evolution was described in a revision of the previous standard, ISO 5347.1, which resulted in the new ISO 16063.11 in 1999. In the new standard, the sine approximation method was introduced. As the input is known to be a sine, a least square fitting of the results calculated from the interferometer output gives good results. The fitting is essentially a filtering at the known frequency. Therefore, with the advanced high-resolution FFT analysers available today, it is logical to use such analysers to do the filtering.

A system following these concepts has been realised. This system will be described, and an evaluation of the system’s sensitivity to different imperfections (e.g., noise and gain differences in the interferometer output) will be reported. The evaluation is made using computer-generated test signals, and the technique can also be used to verify the system whenever needed.

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A method for measuring the normal incidence transmission loss and related acoustical properties of a sample placed in a four-microphone standing wave tube is described.

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While traditional spectral analysis techniques based on Fourier Transform or Digital Filtering provide a good description of stationary and pseudo-stationary signals, they face some limitations when analysing highly non-stationary signals. These limitations are overcome using Time-frequency analysis techniques such as Wavelet Transform, Short-time Fourier Transform, and Wigner-Ville distribution. These techniques, which yield an optimum resolution in the time and frequency domain simultaneously, are described in this article and their advantages and benefits are illustrated through examples.

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Dyn-X Technology, Order Tracking, and Comparison of IBEM PNAH and LDV Methods

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Acoustical Solutions in the Design of a Measurement Microphone for Surface Mounting; Combined NAH and Beamforming Using the Same Array; Patch Near field Acoustical Holography Using a New Statistically Optimal Method

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This article explains the basic principles of Beamforming, including the main performance parameters Resolution and Sidelobe Level. Special attention is given to the influence of array design and to cross-spectral beamforming. Different array designs, including Brüel&Kjær’s newly patented wheel array design, are described and compared, and the basic principle of Brüel&Kjær’s geometry optimisation method is outlined. A new, improved version of cross-spectral beamforming used in Beamforming Software Type 7768 is introduced and its benefits are verified. The article also provides some guidelines for performing good measurements and finally, describes a set of measurements representing typical applications.

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In a previous paper (SENSOR 1997), the principle of a triaxial, piezoelectric accelerometer was described.

This paper describes a similar design but with the piezoelectric ring suspended on only three simple supports. The basic theory behind the principle is explained and measurement results from a very compact, practical unit are also described.

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The Influence of Environmental Conditions on the Pressure Sensitivity of Measurement Microphones; Reduction of Heat Conduction Error in Microphone Pressure Reciprocity Calibration; Frequency Response for measurement microphones – a question of Confidence; Measurement of microphone random-incidence and pressure-field responses and determination of their uncertainties

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This article gives a general introduction to the principles and applications of the Brüel & Kjær Non-stationary STSF measurement technique, with special emphasis on the new functionality included with Version 2.0 of the software package Type 7712. After a general introduction, the principles of Bad Measurement Point Interpolation and Envelope Intensity are descibed. Two case studies of typical applications are then outlined: the first one, which is measurement of brake squeal, is an example of transient noise source localisation. The second case study is mapping and analysis of microphenomena and orders in engine noise radiation.

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In this article the filter characteristics of the Vold-Kalman Order Tracking Filter are presented. Both frequency response as well as time response and their time-frequency relationship have been investigated. Some guidelines for optimum choice of filter parameters are presented. The Vold-Kalman filter enables high performance simultaneous tracking of orders in systems with multiple independent shafts. With this new filter and using multiple tacho references, waveforms, as well as amplitude and phase may be extracted without the beating interactions that are associated with conventional methods. The Vold-Kalman filter provides several filter shapes for optimum resolution and stop-band suppression. Orders extracted as waveforms have no phase bias, and may hence be used for playback, synthesis and tailoring.

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Danish Primary Laboratory of Acoustics (DPLA) as Part of the National Metrology Organisation; Pressure Reciprocity Calibration – Instrumentation, Results and Uncertainty; MP.EXE, a Calculation Program for Pressure Reciprocity Calibration of Microphones

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A simple set-up for quick and purely electrical QC testing of knock sensors is described in this paper. The main goal is to measure the “impedance” frequency response function of the sensors. The advantages of the test procedure described in this paper are as follows: Simple “pseudo” one channel measurement, from the operator’s point of view. Electrical testing, i.e. no need for shakers, power amplifiers, elaborate fixtures etc.. Pulse excitation, thus very fast — in the order of 100ms for one measurement test. No averaging needed. Standard frequency response function display is used, whereby no complex postprocessing is required. The only requirement is a small junction box with the necessary transformer and some electronics. The power supply for the junction box is fed from the analyzer. The testing method can be extended to other types of transducers such as accelerometers.

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While traditional spectral analysis techniques based on Fourier Transform or Digital Filtering provide a good description of stationary and pseudo-stationary signals, they face some limitations when analysing highly non-stationary signals. These limitations are overcome using Time-frequency analysis techniques such as Wavelet Transform, Short-time Fourier Transform, and Wigner-Ville distribution. These techniques, which yield an optimum resolution in the time and frequency domain simultaneously, are described in this article and their advantages and benefits are illustrated through examples.

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The upper frequency limit of a p-p sound intensity probe with a certain microphone separation distance is generally considered to be the frequency at which an ideal probe would exhibit an acceptably small finite difference error in a plane wave of axial incidence. This article shows that the resonances of the cavities in front of the microphones in the usual ‘face-to-face’ configuration give rise to a pressure increase that to some extent compensates for the finite difference error. Thus the operational frequency range can be extended to an octave above the limit determined by the finite difference error, if the length of the spacer between the microphones equals the diameter.

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This article introduces the theory behind digital order tracking, which is implemented by oversampling, interpolation and resampling techniques in Brüel&Kjær instrumentation. As application examples order tacking is applied to the vibration signal from a coast down of a turbogenerator from a power plant station, a run-up of a large marine diesel engine, gated measurement on an automobile engine and finally tracking is combined with time domain averaging on a single cylinder, four stroke engine.

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Ever stricter legislation and competition compel vehicle manufacturers to reduce external vehicle noise. In order to simulate the actual working conditions of vehicles, acoustic testing must be performed not only under stationary but also under non-stationary conditions, e.g. run-up. This article deals with the use of Brüel&Kjær's Spatial Transformation of Sound Fields (STSF) technique as a tool for assessing and understanding the noise radiation phenomena, both qualitatively and quantitatively. Two different implementations of STSF will be described: the well known Cross spectral approach enabling a scan measurement to be applied on a stationary sound source and a new Time domain approach requiring all measurement positions to be acquired simultaneously. While the Cross-spectral STSF is a well established commercially available technique, the Time domain method is still under development.

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This article discusses the errors that are introduced when calculating the impulse response function from the frequency response function using DFT/FFT analysis technique.

This article also represents an expansion of papers presented at the Modal Analysis Conference ISMA 15, held in Leuven, Belgium 1990 and included in the Proceedings of IMAC 9, Firenze, Italy 1991.

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In this article several methods for measurements of damping are summarized with respect to their advantages and disadvantages. Especially the use of Digital Filters (DF) and Fast Fourier Transform (FFT) are compared. In general FFT analysis is best suited for heavily damped structures although with proper memory and postprocessing facilities, lightly damped structures may also be covered, while it is advantageous to use DF analysis when dealing with lightly damped structures only. The use of Time-frequency analysis techniques such as the Wavelet Transform and the Short-time Fourier Transform is also demonstrated.

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The Brüel & Kjær Multi-gas Monitor Type 1302 and Toxic-gas Monitor Type 1306 both utilize interchangeable optical filters to select a band of infrared light at an appropriate wavelength for monitoring some particular gas. This article describes the range of optical filters available: their construction, specification and their selection and use with Type 1302 and/or Type 1306 for a particular monitoring task.

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The Brüel and Kjær Multi-gas Monitor Type 1302 and the Toxic-gas Monitor Type 1306, both used for monitoring gases, are based on photo-acoustic spectroscopy. In photoacoustic spectroscopy (PAS) the gas to be measured is irradiated by intermittent light of pre-selected wavelength. The gas molecules absorb some of the light energy and convert it into an acoustic signal which is detected by a microphone. PAS is an inherently very stable method for detecting very small concentrations of gas. In this article the Brüel & Kjær PAS transducer system is described, the acoustic signal generated by the gas is analyzed, and the influence of noise and other disturbing signals, which are likely to influence the performance of the system, are discussed.

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The Spatial Transformation of Sound Fields (STSF) technique involves a scan using an array of transducers over a (planar) surface close to the source under investigation. From cross spectra measured during the scan, a principal component representation of the sound field is extracted. Any power descriptor of the near field (intensity, sound pressure, etc.) can then be investigated by means of Near-field Acoustic Holography (NAH), while the more distant field can be determined by application of Helmholtz' integral equation. The theoretical foundation of the cross spectrum principal component technique implemented in STSF has been discussed by J. Hald in [1] and the advantages of STSF compared to a number of acoustical holography techniques have been demonstrated. In this article, the practical implementation of STSF is discussed and some applications are described.

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The Spatial Transformation of Sound Fields (STSF) technique involves a scanning over a (planar) surface close to the source under investigation. From cross spectra measured during the scan, a principal component representation of the sound field is extracted. Any power descriptor of the near field (intensity, sound pressure, etc.) can then be investigated by means of Near-field Acoustic Holography (NAH), while the more distant field can be determined by application of Helmholtz' integral equation. The present paper outlines the theoretical foundation of the cross spectral principal component technique implemented in STSF, and relates it to various other NAH techniques. It is demonstrated that compared to these techniques, the cross spectral formulation as implemented in STSF has the advantage that no restrictions are placed on the coherence and bandwidth of the sound field, without requiring simultaneous measurements.

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Part II of the article "Use of Weighting Functions in DFT/FFT analysis" contains the following Appendices referred to in Part I of the article A: Analogy between filter analysis and DFT/FFT analysis,
B: Windows and figures of merit,
C: Effective Weighting of overlapped spectral averaging
D: Experimental Determination of the BT product for FFT-analysis using different weighting functions and overlap,
E: Examples of User Defined Windows,
F: Picket Fence Effect

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This article demonstrates how the analogy between DFT/FFT (Discrete Fourier Transform/Fast Fourier Transform) analysis and filter analysis (analogue or digital) can be used to better understand the applications of different weighting functions used in DFT/FFT. The filter characteristics of the most commonly used weighting functions (also called windows) are illustrated and discussed with respect to their use in various practical applications of system and signal analysis. The mathematical formulations of the analogy as well as rigorous details of the article will be given in the Appendices in Part II of this article to be published in Technical Review No. 4-1987.

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Two-channel FFT analysis is used to evaluate the performance of various accelerometer designs. The measurements demonstrate the importance of using the frequency spectrum for evaluation of the environmental performance of transducers. Furthermore, the superior environmental-performance characteristics of the Briiel & Kjaer Delta Shear® piezoelectric accelerometer design as compared to other designs are demonstrated.

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All rotating machinery generate vibration, the analysis of which renders valuable information about the condition of the machines. With the advent of battery-operated portable FFT Analyzers and Desk-Top Calculators, it is now possible to detect incipient faults in the most common machine elements several months before repair becomes imperative. This article describes the type of vibration signals to be expected for faults in typical elements and the analysis techniques to be used for early detection. Furthermore, diagnostic methods are described which permit estimation of the remaining life of an element once a fault has been detected.

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In this article a practical method is proposed and outlined for determining the Dynamic Capability of Intensity analyzing systems and the Reactivity Index of Intensity measurements. Furthermore, using this method, the amount of error due to phase mismatch, the amount of random error, and the useful frequency range for measuring intensity in different types of sound fields can be determined.

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In many cases, the sound transmission from a speaker to a listener is not perfect, which may result in reduced speech intelligibility. The performance of such a sound transmission system can be quantified by the Modulation Transfer Function: the extent to which the fluctuations in the original signal are preserved in the signal reaching the listener. An illustration is given of the way in which an MTF analysis can be performed and, additionally, how such data are converted into an index which quantifies the effect of a sound transmission system on speech intelligibility.

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An expression for the luminance factor matrix of glass-covered surfaces, such as those used for the Brüel&Kjær reflectance standard, is formulated, and the parameters of the expression are determined, so that a close reproduction of the measured data of the standards is obtained. A compact method is based on the expression, which is well-suited for computer calculations of contrast conditions. Other practical problems of such calculations are considered also, and advice on practicable methods is given.

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The theoretical background of the Hilbert Transform incorporated in the B&K Dual Channel Signal Analyzers Types 2032/2034, is introducing in this article. Using this transform, the normal real valued time domain functions are made complex, which yield two new useful properties - the Envelope and the Instantaneous Frequency. Practical use of the Envelope function is demonstrated.

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In the first part of this article the basic dual channel FFT measurement was introduced and Frequency Response Function estimates and excitation techniques were discussed. In the second part of this article the time domain functions, Impulse Response Function, Autocorrelation and Cross Correlation and their physical interpretation is dealt with in some detail. The implementation of the Hilbert Transform on these time domain functions, to compute the corresponding complex analytical signals, is introduced, and the advantages of using the magnitude in the presentation of these functions in some practical situations are illustrated. Calculation of sound intensity from a dual channel measurement of the sound pressure signals from two closely spaced microphones is discussed in terms of advantages and disadvantages. Formulae for random errors on some of the functions derived from a dual channel measurement on random data are also given.

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The first part of this article introduces basic dual channel FFT measurements. The physical interpretation of the Cross Spectrum, which is the fundamental function in these measurements, and the Coherence Function are dealt with in some detail. Two different methods for estimating the complex Frequency Response Function of a system, from the input and the output signals, are derived, and it is shown which of the two estimates, H1(f) and H2(f), should be used in different practical measurement situations. Various excitation techniques for system analysis are described and their advantages and disadvantages for specific applications outlined. A number of practical measurements, using the Brüel & Kjær Dual Channel Signal Analyzer Type 2032/2034, are presented to illustrate the function estimates obtained with the different techniques. Part 2 of this article deals with the applications of the time domain functions, Hilbert Transform and Sound Intensity.

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Time Delay Spectrometry (TDS) is a relatively new method for measurement of system response. Based on a linear sine sweep it optimizes measurement performance eliminating some earlier drawbacks of swept measurements. By its very nature this method tempts us to adopt an identical complex description in the time and frequency domains. To fully utilize the benefits from this description it is essential to consider the general behaviour of two-port systems and the problems encountered in the measurement of these. The present part of the article is an overview of the theoretical foundation for analysis of linear and time invariant systems. Part II deals more specifically with the TDS technique and its practical implementation.

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In Part I of this article (Technical Review No.3 - 1982) the theoretical concept of sound intensity was described, and the different principles of signal processing were outlined. In part II this article is continued, where the practical aspects of instrumentation are examined such as the requirements that have to be fulfilled in the design of the intensity probe, and how the phase mismatch between the two microphone channels can be eliminated. The signal to noise ratio achievable in the use of the two microphone technique is considered, and the instrumentation required for this technique discussed. Finally, typical applications of sound intensity in the fields of sound power measurement and source localization are also illustrated.

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In this article the theoretical concept of sound intensity (a quantity describing the net flow of acoustic energy) is described, and shows how it makes a distinction between the active and reactive parts of sound fields. Furthermore, it is shown how sound intensity can be measured over a wide frequency range by the use of a specially designed probe, consisting of a pair of closely spaced pressure microphones. The different principles of signal processing are discussed, such as the digital filtering employed in the Bruel & Kjaer Sound Intensity Analyzing System Type 3360, and the use of the Fast Fourier Transform (FFT) technique. The digital filter sound analyzing system operates in real time and calculates third octave spectra more than 100 times faster than an analyzer based upon FFT-technique. The high and low frequency limitations as well as the near field limitations with the two microphone technique are outlined and shown how they are minimised. Calibration procedure is also described. In Part II of the article (Technical Review No.4 - 1982) the practical aspects of instrumentation will be discussed and applications illustrated.