Indirect measurement of acoustic power into a small room at low frequencies

An essential step towards improving sound insulation is a reliable means of quantifying the performance. However, for various reasons sound insulation measurements at low frequencies are associated with relatively high uncertainty and wide variance values. The objective of this research is to develop a method of sound insulation measurement which complements the standard ISO 140 measurement methods by providing improved accuracy at low frequencies. In this paper part of the problem is considered, namely the measurement of power radiated into the receiver room. The ‘peak envelope method’ is based on mode theory and the measurement employs a pair of microphones in the receiver room and a calibrated volume velocity source. No reverberation time measurements are required. The theory is outlined and computer simulations and trial measurements are carried out in order to validate the theory. Good agreement in numerical and experimental validation is demonstrated. We conclude that the peak envelope method is suitable for the measurement of radiated sound power at modal frequencies where ISO 140 methods are poorly adapted. In order to obtain transmission loss, a measure of incident power in the source room will also be required, which will be the subject of future works.     

Monotonic curvature of low frequency decay records in reverberation chambers

In the papers by Larsen [1] and Brüel [2], two interesting problems connected with reverberation room measurements are pointed out and discussed. The first problem is that the ensemble averaged decay curve reveals a monotonic curvature at low frequencies. The second phenomenon is that often systematically larger sound power output values are reported at low frequencies according to the free field method than according to the reverberation room method. In searching for an explanation of these anomalies some measurements and a classical normal mode theory analysis have been made. It is shown that it is not possible to explain fully the curvature of the low frequency decay curves by means of the normal mode theory. The measured curves are more bent than the respective theoretical ones. Most probably, it should be possible to explain this lack of agreement by the fact that the absorption characteristics of normal reverberation chambers significantly deviate from the situation of uniform wall admittance which has been assumed in the theoretical deductions. The theoretical analysis and the comparison between theory and practice indicate that the damping characteristics of the individual waves vary much more than is predicted for a uniform wall admittance. This reasoning is supported by the observation that the monotonic curvature increases when a plane concentrated absorbent is added to one of the walls. One way to decrease the curvature has also been identified. When the room surfaces are provided with randomly placed small samples of low frequency absorbents the resulting decay curves turn out to be almost perfectly linear. Furthermore, it is found that the normal mode theory does not imply significantly different sound power output values than the ISO 3741 model. This fact has been verified with a comparative test. According to the normal mode theory the average sound power output as measured in the reverberant room should equal the free field output. Therefore, one is forced to conclude that the analysis of the classical normal mode theory fails in explaining the anomalies observed.     

压电瞬态响应在换能器声功率测量中的应用

探索一种简便的聚焦超声功率测量方法,利用压电陶瓷片直接接收超声信号,通过机电类比得到压电瞬态响应由压电片在声波作用力下引起受迫振动产生的电压响应与固有振动产生的高频衰减响应叠加而成,分析输出压电信号与换能器声功率之间的换算关系。对输出压电信号进行二次包络提取,获得表征声功率变化的电压幅度曲线,分别找出不同换能器驱动电压下包络曲线的最大峰值电压,将其平方值与声功率计所测声功率进行线性拟合,并对理论关系式中的比例系数进行标定。实验结果所得线性拟合度较高,且标定后所得声功率与声功率计所测值相对误差低于8.7%,证明了通过压电瞬态响应测量换能器声功率具有可行性。      

Sound pressure-level prediction in large rooms containing barriers

A simple-to-use graphical method for estimating the sound pressure level in the shadow zone of a rigid straight-edged barrier for sound radiated from a point source located in a large room is presented and discussed. Also presented is a means based on the graphs introduced for evaluating the noise reduction potential of a barrier in a room or factory work space.     

Determination of sound reduction indices in the presence of flanking transmission

Unlike the standard sound insulation test method, which requires diffuse sound fields in both source room and receiving room, the sound intensity method only requires the source room to be diffuse. Flanking may, in some cases, cause large errors in the sound reduction indices obtained by the standard method, even if the flanking power is measured separately and subtracted from the total power. The sound intensity method allows selective determination of sound reduction indices for test objects surrounded by flanking surfaces, provided the measurement surface in the receiving room is defined in such a way that it encloses nothing but the test object.     

Verbal communication and noise in eating establishments

A new simple prediction model has been derived for the average A-weighted noise level due to many people speaking in a room with assumed diffuse sound field. Due to the feed-back influence of noise on the speech level (the Lombard effect), the speech level increases in noisy environments, and the suggested prediction model gives a 6 dB reduction of the noise level by doubling the equivalent absorption area of the room. This is in contrast to the lowering by 3 dB by doubling of the absorption area for a constant power sound source. The prediction model is verified by experimental data found in the literature. In order to achieve acceptable conditions for speech communication within a small group of people, a guide for the recommended minimum absorption area per person in eating establishments is provided.     

Decay rates and wall absorption at low frequencies

This paper is concerned with evaluating the error of conventional estimates of the boundary absorption of rectangular enclosures, with particular reference to reverberation room sound power measurements. The reverberation process is examined theoretically; the relative contributions to the decay rate from different modes in a rectangular room are calculated from an ensemble average over rooms with nearly the same dimensions. It is shown that the traditional method of determining the absorption of the walls of reverberation rooms systematically underestimates the absorption at low frequencies; the error is computed from the ensemble average. Finally, an unbiased estimate of the sound power radiated by a source in a reverberation room is derived. This estimate involves measurement of the initial decay rates of the room and is, unlike the usual reverberation room sound power estimate, neither based on statistical diffuse field considerations nor on the normal mode theory.     

Prediction model of flow duct constriction noise

The scaling law for aerodynamic dipole type of sound from constrictions in low speed flow ducts by Nelson and Morfey is revisited. A summary of earlier published results using this scaling law is presented together with some new data. Based on this, an effort to find a general scaling law for the sound power for components with both distinct and non-distinct flow separation points are made. Special care is taken to apply the same scaling to all data based on the pressure drop. Results from both rectangular and circular ducts, duct flow velocities from 2 to 120 m/s and sound power measurements made both in ducts and in reverberation chambers are presented. The computed sound power represents the downstream source output in a reflection free duct. In particular for the low frequency plane wave range strong reflections from e.g. openings can affect the sound power output. This is handled by reformulating the Nelson and Morfey model in the form of an active acoustic 2-port. The pressure loss information needed for the semi-empirical scaling law can be gained from CFD simulations. A method using Reynold Average Navier Stokes (RANS) simulations is presented, where the required mesh quality is evaluated and estimation of the dipole source strength via the use of the pressure drop is compared to using the turbulent kinetic energy.     

Walk away method versus reverberation time method in determining the room constant

The room constant is a quantity relating to the sound absorption in an enclosure. It is necessary to room acoustics engineering calculations based on the perfectly diffuse sound field theory, both in noise control work and sound system design. Normally, in the validation of the above-mentioned theory, in situ evaluations of the room constant are performed on the basis of reverberation time measurements in the relevant space. Such methods fail to give accurate results when uncertainties regarding the volumes and areas effectively involved in the reverberation process arise. In this case the walk away method is more advantageous in that it only needs steady state sound pressure level measurements at various distances from a small sound source whose directivity factor is known.In this paper both methods are discussed and experimental results relative to four different rooms are compared.     

A note on the modal kurtosis and the concentration factor in reverberation rooms

The effect known as "weak Anderson localization," "coherent backscattering," or "enhanced back-scattering" is a physical phenomenon that occurs in random systems, e.g., disordered media and linear wave systems, including reverberation rooms: The mean square response is increased at the drive point. In a reverberation room, this means that one can expect an increase of the reverberant sound field at the position of the source that generates the sound field. This affects the sound power output of the source and is therefore of practical concern. The relative increase of reverberant energy is described by the concentration factor, which is usually assumed to be 2. However, because of the stronger direct sound field at the source position, it is obviously very difficult to measure this quantity directly under steady-state conditions. A related parameter of crucial importance for the ensemble statistics of responses in rooms is the modal kurtosis, which is usually assumed to be 3. The modal kurtosis is also very difficult to measure directly. This paper presents the results of an indirect experimental estimation of the two parameters.     

An investigation of simple methods for assessing reverberation time

There is considerable interest in the development of a simple test for sound insulation between dwellings. The assessment of reverberation time is the most difficult part of the procedure to simplify. In this paper six alternative methods are described and evaluated. The first three require no electronic apparatus but are not accurate enough for general use. The fourth involves using sufficient absorbing material in the receiving room to effectively determine the RT. This approach appears worth further development. The fifth approach requires a source of known sound power and the final method employs a simple meter giving a direct reading of the decay time. The last two methods appear to be accurate enough for inclusion in a simplified test method but the simple meter seems to have some advantage.     

Compensation for discarded singular values in vibro-acoustic inverse methods

The paper addresses the inverse problem where source strengths are back-calculated from a sound pressure field sampled at several points. Regularization techniques, such as singular value discarding or Tikhonov regularization, are commonly used to improve estimates of source strength in such situations. However, over-regularization can result in even worse errors. A simple procedure is proposed here to compensate for errors of over-regularization. The basis is to constrain the solution such that the spatial mean of the measured and reconstructed sound pressure are equal. In other words, to set the overall sound power of the equivalent (calculated) sources equal to that of the real source. It is argued that the overall sound power is the most stable and reliable quantity on which to base source strength estimates. Examples of both singular value discarding and Tikhonov regularization are given.     

Possibilities and limits of sound power measurements with a real-time intensity analyzer

The sound power of a number of test objects was determined from spatially averaged intensity measurements. The results show that the influence of room acoustics is insignificant even for rooms of widely different room constants, if the measuring surfaces are exactly defined and if a good space-averaging technique is used. The intensity integrated over a closed surface defining a source-free space compared to the sound pressure integrated over the same surface gives a measure of the capability of a specific intensity measuring system to suppress external noise. For the test arrangements measured with broad band noise, this suppression was found to be 14–18 dB(A). A similar value of 15 dB was found from sound power measurements on a source with high external sound and an analysis of the results in one-third octave bands. From these measurements an analytical function was derived which describes the average error of the spatially averaged intensity as a function of the difference between the external sound level and the source sound level. For practical measurement situations a further analytical function was derived which gives this intensity error as a function of the difference between the measured (spatially averaged) pressure and intensity levels. Thus it is possible to estimate the error of intensity measurements directly from measured intensity and pressure data.     

Sound propagation above an inhomogeneous plane: Boundary integral equation methods

A boundary integral equation method is used to compute the sound pressure emitted by a harmonic source above an inhomogeneous plane. First, the theoretical aspects of the problem (behaviour of the pressure around the discontinuities,…) are studied. Then, a comparison between theoretical levels and experimental levels obtained in an anechoic room is presented. It shows that the boundary integral equation (BIE) method is quite convenient for solving this kind of problem. Two interesting results are pointed out: (i) if only a prediction of maximum sound levels is needed, the attenuation is the same for a cylindrical source, a spherical source and N spherical sources, and so it is possible to transform some three-dimensional problems into two-dimensional ones; (ii) a numerical method of computation of the sound field above an inhomogeneous plane does not provide a correct prediction if each part of the plane is not accurately described by the boundary condition chosen.     

High-power terahertz-wave generation using DAST crystal and detection using mid-infrared powermeter

The exact power output of a table-top-sized terahertz (THz)-wave source using a nonlinear optical process has not been clarified because detectors for these experiments [Si bolometer, deuterated triglycine sulfate (DTGS), etc.] are not calibrated well. On the other hand, powermeters for the mid-infrared (mid-IR) region are well established and calibrated. We constructed a high-power dual-wavelength optical parametric oscillator with two KTP crystals as a light source for difference frequency generation. The obtained powers of dual waves were 21 mJ at ~1300 nm, ten times higher than that of the previous measurement. The device provides high-power THz-wave generation with ~100 times greater output power than that reported in previous works. A well-calibrated mid-IR powermeter at ~27 THz detected the generated THz wave; its measured energy was 2.4 microJ. Although the powermeter had no sensitivity in the lower-frequency range (below 20 THz), the pulse energy at such a low-frequency region was estimated in reference to the output spectrum obtained using a DTGS detector: the energy would be from about the submicrojoule level to a few microjoules in the THz-wave region.     

Performance of time- and frequency-domain binaural beamformers based on recorded signals from real rooms

Extraction of a target sound source amidst multiple interfering sound sources is difficult when there are fewer sensors than sources, as is the case for human listeners in the classic cocktail-party situation. This study compares the signal extraction performance of five algorithms using recordings of speech sources made with three different two-microphone arrays in three rooms of varying reverberation time. Test signals, consisting of two to five speech sources, were constructed for each room and array. The signals were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio of the output. A frequency-domain minimum-variance distortionless-response beamformer outperformed the time-domain based Frost beamformer and generalized sidelobe canceler for all tests with two or more interfering sound sources, and performed comparably or better than the time-domain algorithms for tests with one interfering sound source. The frequency-domain minimum-variance algorithm offered performance comparable to that of the Peissig-Kollmeier binaural frequency-domain algorithm, but with much less distortion of the target signal. Comparisons were also made to a simple beamformer. In addition, computer simulations illustrate that, when processing speech signals, the chosen implementation of the frequency-domain minimum-variance technique adapts more quickly and accurately than time-domain techniques.     

Space variances in the mean-square pressure at the boundaries of a rectangular reverberation room

It was found that it was possible to reduce the number of samples needed for power measurements by sampling at the boundaries in a reverberation room. The space variances in the mean-square pressure of oblique wave fields in a rectangular reverberation room were calculated. The sound source was assumed to be a narrow-band noise source. Space variances throughout the room sigma 2A, on the floor sigma 2B, on the edge sigma 2C, and at the corner sigma 2G were compared with each other numerically. The numerical results confirmed that the relation sigma 2G less than sigma 2C less than sigma 2B less than sigma 2A holds well. Particularly, in a rectangular reverberation room, under the condition that the receiver position is fixed at the corner of the room (corner method), the number of samples reduces to 1/2-1/3 the number of samples needed under the condition that both receiver positions and source positions are changed throughout the room. Some experimental results regarding the power measurements by the corner method are also shown. The experimental results confirmed the suitable estimation of the power for a noise source at low frequencies.     

The choice of frequency weightings and RT correction for measurements in a simple test for sound insulation

A simple form of test for airborne sound insulation could involve generating a broad band pink noise in the source room and measuring the source and receiving room levels with a meter containing frequency weighting networks. In practice, the actual source room spectrum shape will depend on room absorption characteristics and this could lead to an error. The effect of this has been examined by computer simulation. It is also necessary to make some form of RT correction to the receiving room level and two possible forms have been simulated. The results of the simulations have been compared with ISO 717 ratings.     

Specification of absorbed-sound power in the ear canal: application to suppression of stimulus frequency otoacoustic emissions

An insert ear-canal probe including sound source and microphone can deliver a calibrated sound power level to the ear. The aural power absorbed is proportional to the product of mean-squared forward pressure, ear-canal area, and absorbance, in which the sound field is represented using forward (reverse) waves traveling toward (away from) the eardrum. Forward pressure is composed of incident pressure and its multiple internal reflections between eardrum and probe. Based on a database of measurements in normal-hearing adults from 0.22 to 8 kHz, the transfer-function level of forward relative to incident pressure is boosted below 0.7 kHz and within 4 dB above. The level of forward relative to total pressure is maximal close to 4 kHz with wide variability across ears. A spectrally flat incident-pressure level across frequency produces a nearly flat absorbed power level, in contrast to 19 dB changes in pressure level. Calibrating an ear-canal sound source based on absorbed power may be useful in audiological and research applications. Specifying the tip-to-tail level difference of the suppression tuning curve of stimulus frequency otoacoustic emissions in terms of absorbed power reveals increased cochlear gain at 8 kHz relative to the level difference measured using total pressure.