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Uncertainties of Measurements in Room Acoustics
Abstract
The influence of several sources of error on room acoustical measurements have been investigated using maximum-length sequences (MLS). The algorithms for the determination of room acoustical parameters used by different analyzers introduce systematic differences caused by differences in time windowing and filtering, in reverse-time integration and in noise compensation. The overall uncertainty of measurements is of the same magnitude or a little higher than subjectively perceivable changes in room acoustical parameters when the measurements are performed according to ISO/DIS 3382. However, the draft standard allows various procedures to be applied in the processing of impulse responses.
... Regardless of the calculation method, the resulting envelope is logarithmized and the law of its decay is transformed from exponential to linear. The envelope is then checked for thresholds exceeded at minus 5 dB, minus 15 dB, minus 25 dB, and minus 35 dB, after which the corresponding 10 [7], although it is clear that nothing prevents their application when using digital equipment. Due to the use of "time-reversed integration", expression (3) is more convenient for the implementation of 60 T measurements with digital equipment, although, if desired, analog equipment can also be used for calculations. ...
... Other similar examples are given in [9] when overestimated or underestimated values of i T made it difficult to estimate 60 T . Thus, the problem of the optimal choice of the i T parameter is relevant, and a set of studies have been devoted to the search for its solution [10] - [16]. ...
... To date, there are many proposals for choosing the i T value [6] - [16]. It is appropriate to assume that, in addition to obtaining the "correct" value of , i T the possibility of simultaneous estimation of EDT, 10 , ...
The use of voice control of unmanned aerial vehicles is relevant due to the ease of practical use and new opportunities. This technology allows one to simplify the interface, making it more intuitive and natural. However, the quality and intelligibility of speech signals indoors can be significantly impaired by noise and reverberation. Therefore, before using voice technologies, it is desirable to take into account the effect of interferences by preliminary assessment of their parameters. In this paper, an algorithm for estimating the boundary (truncation time) between the informative and non-informative parts of the room impulse response, which allows obtaining believable estimates of the reverberation time, is proposed. The proposed algorithm is two-stage. At the first stage, "rough" envelope of the room impulse response is calculated using the detector-integrator, which allows one to find an approximate value of truncation time and construct an approximate envelope of room impulse response using backward integration method to obtain an approximate estimate of the reverberation time. In the second stage, output data of the first stage are used to refine the truncation time and reverberation time estimates. Experimental tests using recordings of real room impulse responses testify to the efficiency of the proposed algorithm.
... The increasing availability of digital measurement technology and the desire to modernize the established measurement standards (ISO 3382, 1975) to recognize these developments have triggered numerous key investigations in the 1990s. At the same time, adding new concepts required some groundwork to confirm the new contents' significance and explanatory power (e.g., J. S. Bradley (1996); Lundeby, Vigran, Bietz, and Vorländer (1995); Pelorson et al. (1992)). ...
... In a next step, the investigation of Lundeby et al. (1995) picks up a discussion voiced earlier by Barron (1984) and studies how reverberation times can be calculated while ensuring the background noise does not disturb the analysis. Lundeby et al.'s algorithm has attracted some recent attention (Guski & Vorländer, 2015;Jankovic, Ciric, & Pantic, 2016) where it is discussed that different strategies suffice with ISO 3382-1 (2009) but inhere different potential of error and uncertainty. ...
... Clarity The starting point for the uncertainty discussion of the clarity metric is its definition based on the works of Reichardt et al. (1974), its standardization in ISO 3382-1 (2009) and its equivalent for discretely sampled impulse responses p(s i ) as shown in Equation 6.2.9. Lundeby et al. (1995) add a compensation term to recognize the influence of background noise. Comparable approaches to tackle the noise problem may exist, but this reference is intended to point to a specific definition of an algorithm to avoid making the uncertainty discussion ambiguous. ...
Regardless of the field, measurements are essential for validating theories and making well-founded decisions. A criterion for the validity and comparability of measured values is their uncertainty. Still, in room acoustical measurements, the application of established rules to interpret uncertainties in measurement is not yet widespread. This raises the question of the validity and interpretability of room acoustical measurements.
This work discusses the uncertainties in measuring room acoustical single-number quantities that complies with the framework of the ``Guide to the Expression of Uncertainty in Measurement''(GUM). Starting point is a structured search of variables that potentially influence the measurement of room impulse responses. In a second step, this uncertainty is propagated through the algorithm that determines single-number quantities.
A second emphasis is placed on the investigation of spatial fluctuations of the sound field in auditoria. The spatial variance of the sound field in combination with an uncertain measurement position marks a major contribution to the overall measurement uncertainty. To reach general conclusions, the relation between changes in the measurement location and the corresponding changes in measured room acoustical quantities is investigated empirically in extensive measurement series.
This study shows how precisely a measurement position must be defined to ensure a given uncertainty of room acoustical single-number quantities. The presented methods form a foundation that can be exibly extended in future investigations to include additional influences on the measurement uncertainty.
... Therefore, the dynamic range of the EDC is insufficient for the reliable determination of RT. To overcome the problem, studies and achievements were proposed to mitigate insufficient dynamic ranges in the Schroeder decay functions [10][11][12][13][14][15]. The typical research emphasized truncating the RIR at a time point to generate noise-free decay curves [12,13]. ...
... The error of RT determination caused by improper truncation depends on the TT displacement. Thus, Lundeby et al. [10] proposed a correction term to prevent the effects. The correction part assumes an exponential decay, which should be the same as the decay slope of the estimation range (ER) ranging from the TT to a certain point [9,10]. ...
... Thus, Lundeby et al. [10] proposed a correction term to prevent the effects. The correction part assumes an exponential decay, which should be the same as the decay slope of the estimation range (ER) ranging from the TT to a certain point [9,10]. However, this range has been influenced by noise since the signal and noise energy are equal at the TT. ...
The noise effects in the room impulse response (RIR) make the decay range of the integrated impulse response insufficient for reliable determination of reverberation time (RT). One of the preferred techniques to minimize noise effects is based on noise subtraction, RIR truncation, and correction for the truncation. The success of RT estimation through the method depends critically on the accurate estimation of the truncation time (TT). However, noise fluctuation and RIR irregularities can lead to discrepancies in the determined TT from the optimal value. The general goal of this paper is to improve RT estimates. An iterative procedure based on a non-exponential decay model consisting of a double-slope decay term and a noise term is presented to estimate the TT accurately. The model parameters are generated until the iterative procedure converges to a minimum difference between the energy decay curve (EDC) generated by the model and the Schroeder decay function. The decay rates of the EDCs with added pink noise levels are compared to those of the EDCs with low background noise. In addition, the detected TTs and the corresponding RTs are compared with the existing method and the noise compensation method (subtraction–truncation–correction method).
... A well-known and widely used method to calculate RT is determined by the energy decay curve (EDC) generated by Schroeder's method [6]; however, the measured room impulse response (RIR) presents ambient noise, and equipment noise may deteriorate the EDC, leading to errors in predicting room acoustic parameters [7,8]. The relative errors for RTs, early decay time (EDT), and other acoustic parameters, without noise compensation, could exceed 5% [9,10]. ...
... This problem identified two casual factors: the upper integration time of RIR and the estimated background noise levels at the truncation time. Consequently, a correction term was added to the integration to prevent the truncation error [9,13,14]. For mathematical advantages, nonlinear regression methods [15] were investigated to fit the RIR to calculate the slope of the EDC. ...
... The three rooms enabled tranquil environments where the noise levels estimated from the RIRs were less than −70 dB. The values of the noise levels were estimated from the normalized energy time curve (normalized energy RIR) determined using iterative techniques [9]. Once the sets of the factors were achieved, the RIRs measured in two normal rooms (room A and room B), listed in Table 1, were used to verify the accuracy of the obtained optimal factors of the GBSS algorithm (Section 4.4). ...
The generalized spectral subtraction algorithm (GBSS), which has extraordinary ability in background noise reduction, is historically one of the first approaches used for speech enhancement and dereverberation. However, the algorithm has not been applied to de-noise the room impulse response (RIR) to extend the reverberation decay range. The application of the GBSS algorithm in this study is stated as an optimization problem, that is, subtracting the noise level from the RIR while maintaining the signal quality. The optimization process conducted in the measurements of the RIRs with artificial noise and natural ambient noise aims to determine the optimal sets of factors to achieve the best noise reduction results regarding the largest dynamic range improvement. The optimal factors are set variables determined by the estimated SNRs of the RIRs filtered in the octave band. The acoustic parameters, the reverberation time (RT), and early decay time (EDT), and the dynamic range improvement of the energy decay curve were used as control measures and evaluation criteria to ensure the reliability of the algorithm. The de-noising results were compared with noise compensation methods. With the achieved optimal factors, the GBSS contributes to a significant effect in terms of dynamic range improvement and decreases the estimation errors in the RTs caused by noise levels.
... It has to be noted that this method requires the truncation time to be sufficiently high for the resulting error to be negligible in the desired evaluation range. 39,41 Lundeby et al. 41 state this criterion to be fulfilled for times before decay levels of 20dB above the noise level at the intersection time. Considering directionally varying decay rates, a joint valid evaluation range for all directions has to be chosen which is prescribed by the shortest intersection time over all directions. ...
... It has to be noted that this method requires the truncation time to be sufficiently high for the resulting error to be negligible in the desired evaluation range. 39,41 Lundeby et al. 41 state this criterion to be fulfilled for times before decay levels of 20dB above the noise level at the intersection time. Considering directionally varying decay rates, a joint valid evaluation range for all directions has to be chosen which is prescribed by the shortest intersection time over all directions. ...
... 48 The DEDCs were calculated using Schroeder integration as detailed in equation (4) with an integration limit corresponding to the intersection time and noise power determined using Lundeby's method. 41 The resulting DEDCs were subsequently truncated to the minimum out of the time corresponding to an energy decay of 65dB in the omnidirectional EDC, calculated from the zero-order SH-DRIR, and the time according to the joint valid decay range criterion specified in the previous section. ...
The analysis of the spatio-temporal features of sound fields is of great interest in the field of room acoustics, as they inevitably contribute to a listeners impression of the room. The perceived spaciousness is linked to lateral sound incidence during the early and late part of the impulse response which largely depends on the geometry of the room. In complex geometries, particularly in rooms with reverberation reservoirs or coupled spaces, the reverberation process might show distinct spatio-temporal characteristics.
In the present study, we apply the analysis of directional energy decay curves based on the decomposition of the sound field into a plane wave basis, previously proposed for reverberation room characterization, to general purpose performance spaces. A simulation study of a concert hall and two churches is presented uncovering anisotropic sound field decays in two cases and highlighting implications for the resulting temporal evolution of the sound field diffuseness.
... where L is called upper limit of integration (ULI). The ULI considerably influences the resulting EDF due to the background noise inherent in room-acoustic measurements [50,8], thus prompting many studies on its optimal choice [51,52,53]. Figure 2.1 illustrates different graphical representations of sound-energy decay in rooms. ...
... For EDFs with dynamic ranges smaller than 60 dB, a decay of 20 dB or 30 dB can be evaluated and extrapolated to 60 dB instead, corresponding to the measures T 20 and T 30 , respectively [49,17]. Noise subtraction [55,53] or RIR truncation [51,52,53] mitigate the influence of background noise on the RT estimate. Alternatively, nonlinear models with a dedicated noise term can fit the entire EDF including background noise to obtain unbiased RT estimates T [56, 57]. ...
The study of room acoustics has traditionally been of interest in architectural planning and design. With the spread of virtual- and augmented-reality technology, room-acoustic modelling has also become increasingly relevant for audio engines. The dynamic and fast-paced nature of such applications requires rendering systems to operate in real-time. However, accurate state-of-the-art room-acoustic-simulation technology is often computationally expensive, limiting its use for audio engines. Data-driven methods offer the potential to bypass expensive simulations, while ensuring convincing perceptual experiences. This dissertation works towards data-driven audio engines by exploring the interaction between room-acoustic modelling and data-driven methods. It comprises five peer-reviewed publications that investigate automatic data acquisition, robust room-acoustic analysis in complex environments, and data-driven room-acoustics rendering.
As sound propagates through a room, it interacts with various surfaces, leading to a gradual energy decay over time. The properties of this energy decay significantly influence the acoustic impression evoked by a room, making it a widely studied topic in room-acoustic research. The first part of this thesis provides an overview of sound-energy decay, its analysis, and challenges associated with complex geometries featuring multiple rooms and non-uniform absorption-material distributions. To this end, it introduces a neural network for multi-exponential sound-energy-decay analysis. Moreover, spatial and directional variations of sound-energy decay are investigated, and a compact representation to model them is proposed.
The second part of this thesis is centred around data-driven methods and explores how they can be applied to room-acoustics research. After elaborating on the properties of room-acoustic data, techniques for its large-scale acquisition are investigated. Two of the contained publications describe autonomous robot systems for conducting room-acoustic measurements. While the first one describes the general idea and the design constraints of a practical system, the second one extends the measurement strategy to complex geometries featuring multiple connected rooms. An overview of commonly used machine-learning concepts is provided, focusing on the ones relevant for the included publications. Finally, several applications of data-driven methods in room-acoustics research are described, including a summary of a late-reverberation rendering system proposed in one of the appended publications.
... The excitation signal had a length of 2 17 samples and was generated at a sampling rate of 44.1 kHz. The room acoustic measurements were performed and evaluated aiming at conformity with ISO 3382-2 [10] by optimising the placement of the source and the receivers, applying noise detection and compensation [18]. Meeting standard measurement precision level, the two and three source and receiver positions were measured, respectively, with two averages per combination. ...
... All room acoustic parameters presented in Figure 5 were calculated per source-receiver combination [11,27], applying noise detection and compensation [18], and subsequently arithmetically averaged across combinations per frequency band. The optimisation measures effectively decreased the RT in the lowest evaluated octave band by about 0.4 s. ...
The Institute for Hearing Technology and Acoustics of RWTH Aachen University owns a number of acoustically optimised laboratories and specialised hardware, allowing the conduction of various acoustic measurements and perceptual experiments. Some of these laboratories have been improved and adapted, others were conceptualised and implemented completely from scratch, to develop the results presented in the
dissertation “Spatial audio reproduction for hearing aid research: System design, evaluation and application”. The key specifications of the laboratory infrastructure and hardware used are presented in this report, including compact acoustic evaluations and descriptions of the reproduction environments, spatial audio reproduction systems, and systems to measure generic and individual receiver directivities.
... Truncation at intersection time t i First, one can try to find the intersection of the signal energy decay and the noise iteratively with the Lundeby algorithm [10] and cut off the EDC directly at t i (3). This method is part of ISO 3382-1 [1]. ...
... In a second step, the truncated part of the EDC can be compensated with the ernergetic term C comp gained by a linear regression according to [10] (4). This method is also mentioned in the ISO 3382-1 standard [1]. ...
Nowadays, the use of quadcopter UAVs is becoming increasingly popular in a wide range of civil applications. Another promising application for quadcopter UAVs would be to perform automated room acoustic measurements with high spatial resolution. Obviously, the flight noise of a quadcopter UAV has a considerable influence on the quality of a room acoustic measurement.This contribution presents a feasibility study regarding the determination of room acoustic parameters by room impulse response measurements under the influence of flight noise from a quadcopter UAV. For this purpose, directional flight noise radiation characteristics of the quadcopter UAV are determined in a first step. On the basis of the directional radiation characteristics, an optimal position of the measurement microphone/s to be placed on the quadcopter UAV can thus be determined. In a second step, existing room impulse responses are subjected to flight noise from the quadcopter UAV and noise compensation techniques are validated using these synthesized data. Finally, room acoustic measurements are performed with and without present flight noise from the quadcopter UAV. These measurements are used to evaluate the quality of the room acoustic measures as a function of the noise present due to the quadcopter UAV.
... Background and system noise in the down-sampled p 2 -response was then corrected using a technique proposed by Lundeby et al. [65]. The p 2 -response was first converted to decibels so that the acoustic decay portion of the recording was approximately linear. ...
... The validity of room acoustics simulations is an on-going area of research with periodic round-robin comparisons of simulation algorithms and practices (Vorländer, 1995). The fact that, in this work, computer simulations were no more successful than a Sabine reverberation time prediction indicates that the effect of state of the diffusion is not correctly incorporated into current computer models (although surprisingly variations in the NSDRT were reproduced). ...
In this research, indicators of diffusion appropriate for the analysis of auditoria are discussed along with the influence which room configuration, such as the presence of seating, scattering surfaces and balconies, have on the state of diffusion. Two indirect indicators of diffusion have been analysed: the Normalised Standard Deviation
of Reverberation Time (NSDRT) as proposed by Davy (1979) and the Normalised Standard Deviation of Level (NSDL) which is proposed here and based on work by Chiles and Barron (2004). Both the NSDRT and NSDL can be evaluated from room impulse responses.
To investigate the state of diffusion present in auditoria, impulse responses have been recorded in a 1:25 scale model rectangular concert hall. In this environment, it has been found that the NSDRT responds primarily to changes in sound field isotropy, the NSDL to changes in homogeneity. Taken together, a good judgement
of the overall degree of sound field diffusion can be made.
Based on measurements of the NSDRT and NSDL, it has been shown that in a rectangular auditorium, due to the presence of absorbing audience seating, one-and two-dimensional reverberant fields may become established in the upper part of the room volume. Scattering surfaces and balconies can disturb such one- and two-dimensional reverberant fields but, to be effective, scattering treatments must
be applied to surfaces perpendicular to the plane of reverberation. The application of scattering surfaces to the side walls can though impede the transmission of acoustic energy along the auditorium by reflection and lead to deficient sound levels in rear positions. The only location for the application of scattering treatments which results in increases in both sound field isotropy and homogeneity is the stage region.
It has also been demonstrated that the accuracy of both Sabine and computer predictions of reverberation time are highly dependent on the degree of diffusion. Computer models have been shown to reproduce the variations in NSDRT observed in scale model measurements.
... There are still known factors impacting the reliability 74 Chapter III. Auralization usage survey (through plausibility) of the results such as having proper anechoic material and precise associated acoustic measurements (Lundeby et al., 1995;Pätynen et al., 2008). ...
... Reliability of both the technology and the results remains important: the better the renderings, the more immersive the simulation, the more people engage themselves in the project, as reported by an interviewee from the survey. While there are known factors impacting the reliability of the results such as having proper anechoic material, linked to the precision of the associated acoustic measurements, or the variability of simulation algorithms (Lundeby et al., 1995;Vorländer, 2013b;Brinkmann et al., 2019), the tools themselves also need to be reliable and stable, which is impeded by the rapid evolution of technologies. These difficulties linked to the frequent updates of tools and softwares proved to be one of the main difficulties. ...
This thesis investigated the use of auralization in the design phase of architectural projects. While this technology, which consists of rendering audible numerical acoustical simulations, has been extensively used in research, from cognitive to human-computer interfaces to archeology to concert halls acoustics evaluation studies, only limited data existed on its use by acoustical consultants for acoustical design. The question was to evaluate if auralizations can improve the processes of acoustical design for the conception of architectural spaces. Based on the practical acceptability theory, the use of auralizations has been studied through questionnaire and interviews of acoustical consultants, as well as the observation of a practical case study project, conducted in collaboration with Theatre Projects Consultants. These enabled the identification of the main uses of auralizations, as well as the difficulties encountered that impede the adoption of the technology. One of the requirements for its adoption is the accuracy and reliability of both the results and the tools themselves. Therefore, the stability of auditory perception was assessed in perceptive studies along a set of subjective attributes, with auralizations rendered over different interfaces, including sound reproduction methods and VR visual interfaces. The two sound reproduction methods compared were head-tracked Binaural and Ambisonic while the VR visual devices compared were a HMD and a CAVE-light system. A relative stability of auditaptory perception was observed, while the perceived Envelopment and Apparent Source Width were slightly impacted by the rendering system in both cases. Efforts in the adaptability of auralization tools for use in architectural design and knowledge transfer from research to industry are still needed for a better integration of auralizations in architects and acousticians workflow.
... The room acoustic parameters included in the development of the app so far are reverberation time (RT20, RT30), bass ratio, clarity (C50, C80), definition (D50, D80), center time [1] and the speech transmission index (STI male , STI female ) [3]. Their calculation is based on a room excitation by impulsive stimuli and is supported by a number of features such as an automated start-and endpoint truncation of room impulse responses as well as energy compensation for the truncation and an iterative SNR assessment [4]. An octave and third-octave band IIR filter bank [2] for signal analysis was implemented, as well as the option to use time reverse and zero phase filtering [5] for narrow filter bandwidths and short reverberation time measurements. ...
... A mean impulse response was calculated and filtered in octaves. An endpoint truncation according to the algorithm proposed by Lundeby et al. [4] was done without truncation energy compensation. Subsequently the reverberation time RT20 was calculated by the presented applications algorithms and the two aforementioned toolboxes. ...
... enhances the dynamic range[8]. Subsequently it combines the Ambisonics upmix and equalization algorithms by Zaunschirm et al.[6, appendix B] with the roughness correction of Amengual Garí et al. [9, section 4.2.2]. ...
In order to examine the impact of different architectural features on sound source distance perception in future studies, the uncertainty of auralization results needs to be minimized and quantified. The RAVEN room acoustic software was benchmarked against measurements of the loudspeaker orchestra by Pätynen et al. in the Gasteig concert hall in Munich. The source arrangement used is essential as it diversifies the direct and reflected sound paths, changing the perception compared to a single omnidirectional room excitation. Various objective and subjective measures are presented to illustrate the particularities identified during our research.
... In article [13], it is recommended to avoid finding the value T i at all, subtracting the mean square of the background noise from h 2 (t). Although this idea was supported in [14], later the same authors [15] proposed an iterative algorithm for estimating T i , the idea of which is to perform calculations according to (1) with different values of the averaging interval T. The mentioned algorithm is quite complicated for practical implementation. In addition, its resistance to background noise has not been studied sufficiently. ...
The task of estimating the reverberation time is relevant in the acoustic examination of premises for both civilian purposes (kindergartens, educational institutions, concert halls, etc.) and military purposes (control centers). Reverberation time measurement is usually performed by the method of inverse integration of the room impulse response. However, due to the existence of background noise, the problem of choosing the moment of time (truncation time) from which integration should begin arises. In this paper, the accuracy of the algorithm for calculating the reverberation time, where the truncation time is defined as the moment of approach to zero of the derivative of the "squaring-moving averaging" system output signal, is studied. Estimates of the bias, standard deviation, and total error for the values of the signal-to-noise ratio and reverberation time typical for classrooms are obtained. At a signal-to-noise ratio of 45 dB, for measurements in a wide frequency band of 80 Hz-11 kHz, the total relative error of the reverberation time estimation does not exceed 6% for reverberation time values of 0.6-1.2 s. When measuring reverberation time in octave frequency bands, the error reaches 20% for the frequency band in the vicinity of 125 Hz, decreases to 10% for the frequency band in the vicinity of 500 Hz, and does not exceed 3% for the frequency band in the vicinity of 8 kHz.
... [6] para calcular el mejor punto de inicio. La Figura 5 también muestra el comportamiento de los parámetros acústicos obtenidos mediante las simulaciones en I-SIMPA. ...
En este estudio se selecciona un espacio físico con geometría rectangular y se realiza un análisis comparativo entre los parámetros acústicos generados a partir de la medición in situ de sus respuestas impulsionales, utilizando un parlante unidireccional y los parámetros acústicos obtenidos a partir de métodos numéricos en un programa de simulación, donde se utiliza un parlante omnidireccional como fuente de sonido. Se evalúa también el posible beneficio de aproximar una fuente sonora onmidireccional mediante la rotación de una fuente unidireccional en el plano horizontal. Los resultados obtenidos indican que las mediciones realizadas con el parlante unidireccional permiten obtener una caracterización general de la acústica del espacio, con niveles de error que podrían ser tolerados, sobre todo a frecuencias medias, con la ventaja de tener un costo menor.
... The increasing availability of digital measurement technology and the desire to modernize the established measurement standards (ISO 3382, 1975) have triggered numerous key investigations in the 1990s. At the same time, adding new concepts required some groundwork to confirm the new contents' significance and explanatory power (e.g., J. S. Bradley (1996); Lundeby, Vigran, Bietz, and Vorländer (1995); Pelorson et al. (1992)). To this present day, many users developed a rich experience using this standard. ...
Measurements are essential for validating theories and making well-founded decisions. A criterion for the validity and comparability of measured values is their uncertainty. The latter aspect is not always applied in room acoustics. Using the standardized "Guide to the Expression of Uncertainty in Measurement" (GUM) framework, this contribution discusses the uncertainties of room impulse response measurements and the calculation of room acoustical single-number quantities. A further emphasis is placed on the investigation of spatial fluctuations of the sound field in auditoria. The influence of an uncertain measurement position on the overall measurement uncertainty is discussed. The discussion leads to a hierarchical ranking of contributions that affect the uncertainty in room acoustical measurements and indicates where to direct efforts to reduce uncertainty. The presented methods form a foundation that can be flexibly extended in future investigations to include additional influences on the measurement uncertainty.
... Therefore, performing those measurements with the highest possible accuracy is essential. However, it has been proven that in most acoustic measurements, significant uncertainty is present [1][2][3]. The direct connection between the source orientation and the source-receiver configuration's influence on the results of measurements was proven [2,4]. ...
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The presented research describes the method of so-called coupled speakers used in the room acoustic measurements. It is a practical and cheap alternative to commonly used dodecahedral loudspeakers, which may be essential for multiple professionals and laboratories working in room acoustic. The solution offers proper results compared to dodecahedral sound sources for room acoustic parameters measurements, such as EDT, T30, D50, and C80, in the range of 250–2000 Hz.
Abstract
Omnidirectional sources used in room acoustics usually take the form of multi-speaker sources. Few alternatives for the most commonly used dodecahedral sound source have been derived recently. The project aimed to measure room acoustic parameters using three different sound sources: a dodecahedron, a cube, and a new source of two coupled loudspeakers. The measurements were made by rotating the sources every 15 degrees. The differences in the EDT, T30, D50, and C80 parameters in the function of the rotation angle of the sources were analyzed. Statistical analysis was carried out to examine the sensitivity of the measured parameters’ JND (just a noticeable difference) on the source’s rotation angle. This presentation will show the results and analysis of measurements showing the influence of the used source on obtained parameters and the validation of coupled speakers’ use. A comprehensive discussion of the results obtained with different sources (coupled, dodecahedral, cubic) will be provided. The results confirmed using the coupled speakers as an alternative for omnidirectional sound source in the range of 250–2000 Hz.
... In particular, for what concerns the computing of the monoaural parameters on the 3OA IR recordings, an iterative routine was scripted that computed the average and the standard deviation for each octave band frequency by taking for each church the IR in all receiver positions, extrapolating the related omnidirectional channel tracks, and passing them to the ita_roomacoustics() function. Notably, before computing the room acoustics parameters, the ita_roomacoustics() function truncates the IR at the intersection time, to cut out the noise at the IR tail detected through the Lundeby-Algorithm [11], and applies correction for the truncation in accordance with the directions of the ISO 3382-1 [2]. ...
... The increasing availability of digital measurement technology and the desire to modernize the established measurement standards (ISO 3382, 1975) have triggered numerous key investigations in the 1990s. At the same time, adding new concepts required some groundwork to confirm the new contents' significance and explanatory power (e.g., J. S. Bradley (1996); Lundeby, Vigran, Bietz, and Vorländer (1995); Pelorson et al. (1992)). To this present day, many users developed a rich experience using this standard. ...
Measurements are essential for validating theories and making well-founded decisions. A criterion for the validity and comparability of measured values is their uncertainty. The latter aspect is not always applied in room acoustics. Using the standardized "Guide to the Expression of Uncertainty in Measurement" (GUM) framework, this contribution discusses the uncertainties of room impulse response measurements and the calculation of room acoustical single-number quantities. A further emphasis is placed on the investigation of spatial fluctuations of the sound field in auditoria. The influence of an uncertain measurement position on the overall measurement uncertainty is discussed. The discussion leads to a hierarchical ranking of contributions that affect the uncertainty in room acoustical measurements and indicates where to direct efforts to reduce uncertainty. The presented methods form a foundation that can be flexibly extended in future investigations to include additional influences on the measurement uncertainty.
... The IR of each of the twelve transfer paths between the respective microphone and loudspeaker position is calculated by direct deconvolution of the recorded signal with the exponential sinusoidal sweep. The Peak Signal-to-Noise-Ratio P SN R(f m ), according to Lundeby [43], averaged over all IRs in the relevant frequency range (20 Hz ≤ f m ≤ 200 Hz) is shown in fig. 5. Subsequently, the IRs are cropped to 30 s and transformed in the frequency domain to obtain the twelve TFs with a frequency resolution of ∆f ≈ 0.03 Hz. Since both the subwoofer and the dodecahedron loudspeaker do not have a linear amplitude frequency response (cf. ...
Porous acoustic absorbers have excellent properties in the low-frequency range when positioned in room edges, therefore they are a common method for reducing low-frequency reverberation. However, standard room acoustic simulation methods such as ray tracing and mirror sources are invalid for low frequencies in general which is a consequence of using geometrical methods, yielding a lack of simulation tools for these so-called edge absorbers. In this article, a validated finite element simulation model is presented, which is able to predict the effect of an edge absorber on the acoustic field. With this model, the interaction mechanisms between room and absorber can be studied by high-resolved acoustic field visualizations in both room and absorber. The finite element model is validated against transfer function data computed from impulse response measurements in a reverberation chamber in style of ISO 354. The absorber made of Basotect is modeled using the Johnson-Champoux-Allard-Lafarge model, which is fitted to impedance tube measurements using the four-microphone transfer matrix method. It is shown that the finite element simulation model is able to predict the influence of different edge absorber configurations on the measured transfer functions to a high degree of accuracy. The evaluated third-octave band error exhibits deviations of 3.25dB to 4.11dB computed from third-octave band averaged spectra.
... Several approaches were proposed to counter the effect of noise on a) georg.gotz@aalto.fi b) Also at: Media Lab, Department of Art and Media, Aalto University, Otakaari 5, 00250 Espoo, Finland. the reverberation time estimation 3,5,7,8 . Alternatively, Xiang 9 and Karjalainen et al. 10 proposed to include an additional noise term in the model and perform a nonlinear regression. ...
An established model for sound energy decay functions (EDFs) is the superposition of multiple exponentials and a noise term. This work proposes a neural-network-based approach for estimating the model parameters from EDFs. The network is trained on synthetic EDFs and evaluated on two large datasets of over 20000 EDF measurements conducted in various acoustic environments. The evaluation shows that the proposed neural network architecture robustly estimates the model parameters from large datasets of measured EDFs, while being lightweight and computationally efficient. An implementation of the proposed neural network is publicly available.
... is estimated from the source signal s(t) and the observed (target) signalŷ(t). Per [29], the final ten percent of the impulse response is assumed to be noise-dominated. While the spectrum ofĥn does not contain the true spectrum |N (ω)| of the additive noise in the system, we obtain an estimate for |N (ω)| by multiplication of the spectrum of the noise-dominated portion ofĥn with S(ω). ...
Impulse response estimation in high noise and in-the-wild settings, with minimal control of the underlying data distributions, is a challenging problem. We propose a novel framework for parameterizing and estimating impulse responses based on recent advances in neural representation learning. Our framework is driven by a carefully designed neural network that jointly estimates the impulse response and the (apriori unknown) spectral noise characteristics of an observed signal given the source signal. We demonstrate robustness in estimation, even under low signal-to-noise ratios, and show strong results when learning from spatio-temporal real-world speech data. Our framework provides a natural way to interpolate impulse responses on a spatial grid, while also allowing for efficiently compressing and storing them for real-time rendering applications in augmented and virtual reality.
... The works by Chu [36] and Hirata [37] have reported the relevance of the post-processing method used for the calculation of the acoustic parameters. Furthermore, Lundeby et al [38] showed that the determination of room acoustical parameters could depend on the algorithms utilized, which introduce systematic differences caused by differences in noise compensation, time-windowing and filtering, reversetime integration and materials properties [39]. Actually, only a few standards consider the importance of thermo-hygrometric conditions during acoustic (especially noise) measurements or propagation. ...
... This slope can then be used to derive the time it takes for each frequency band of the measured RIR to drop 60 dB SP L from its initial energy. For this system, the process uses octave frequency band separation followed by RIR-starting point computation, curve-fitting and Lundebyiteration [25] for each frequency band, all mainly based on the AcMus -Room Acoustic Parameters MATLAB Toolbox [26]. Using this methodology we derived mean representations of octave-frequency band T 60 for each of the trained room categories and then used the parametric BRIR synthesis to generate a generic BRIR that can achieve plausible results across multiple instances of the same room category. ...
... Investigations carried out by statistically elaborating subjective judgments of many listeners have shown that, beyond the subjective evaluations, there are some objective parameters of the acoustic field capable of acoustically characterizing the spaces. Over the years, numerous scholars have proposed various methods of determining these parameters, assigning each of them a different weight [7][8][9][10][11]. ...
The measurement of reverberation time is an essential procedure for the characterization of the acoustic performance of rooms. The values returned by these measurements allow us to predict how the sound will be transformed by the walls and furnishings of the rooms. The measurement of the reverberation time is not an easy procedure to carry out and requires the use of a space in an exclusive way. In fact, it is necessary to use instruments that reproduce a sound source and instruments for recording the response of the space. In this work, an automatic procedure for estimating the reverberation time based on the use of artificial neural networks was developed. Previously selected sounds were played, and joint sound recordings were made. The recorded sounds were processed with the extraction of characteristics, then they were labeled by associating to each sound the value of the reverberation time in octave bands of that specific room. The obtained dataset was used as input for the training of an algorithm based on artificial neural networks. The results returned by the predictive model suggest using this methodology to estimate the reverberation time of any closed space, using simple audio recordings without having to perform standard measurements or calculate the integration explicitly.
... The papers by Chu [27] and Hirata [28] have shown the relevance of the post-processing method chosen for the calculation of the acoustic parameters. Moreover, Lundeby et al. [29] showed that the algorithms for the determination of room acoustical parameters introduce systematic differences caused by differences in time-windowing and filtering, reverse-time integration, and noise compensation. ...
The variation of the room acoustic parameters measured in an auditorium is influenced by many variables such as the equipment, the operator, the position and orientation of the sound source and the microphones, and the post-processing method used for the calculation. An influence of fundamental importance is due to the thermo-hygrometric variables, which is commonly neglected. In this article, an experimental analysis concerning the influence of the temperature, relative humidity, and air velocity on acoustic parameters is presented. Thermo-hygrometric variables have been varied and the variation of several room acoustic parameters has been analized. A statistical analysis of the correlations has been obtained and used for the evaluation of the variation of the acoustic parameters by changing the thermo-hygrometric variables. Finally, a statistical analysis has been conducted to find correlations between room acoustic parameters and thermo-hygrometric parameters.
... Zehner et al. [25] reviewed prior interlaboratory studies on reverberation time measurements. Lundeby et al. [21] conducted a study with seven independent participants. However, they focused on MLS measurements and reported only octave bands 125, 1000, and 4000 Hz. ...
International standards ISO 3382-2 and ISO 3382-3 are increasingly applied to determine the room acoustic conditions in open-plan offices because attention to noise control in such workplaces has increased. The precision of those standards in open-plan offices has not been published. The purpose of this study was to determine the precision of ISO 3382-2:2008 and ISO 3382-3:2012 in an open-plan office following the instructions of ISO 5725 standard. Furthermore, the results were analyzed in the light of ISO CD 3382-3:2020 involving improvements related to uncertainty. An accuracy experiment (a.k.a Round-Robin test, inter-laboratory test, intercomparison test) was arranged where nine independent participants conducted the measurements in the same open-plan office. For ISO 3382-3:2012, the reproducibility standard deviations were 530.5 dB, 1.3 dB, 1.3 dB, and 18% for spatial decay rate of speech (D2,S), A-weighted SPL of speech at 4 m distance (Lp,A,S,4m), A-weighted SPL of background noise (Lp,A,B), and distraction distance (rD). The corresponding values for ISO CD 3382-3:2020 were 0.3 dB, 1.1 dB, 0.6 dB, and 16%, respectively. The reproducibility standard deviations for reverberation time were 4.5–16% within 1/1-octave bands 125–8000 Hz. The measurement uncertainty of ISO CD 3382-3:2020 is smaller than that of ISO 3382-3:2012. The values depended mainly on between-laboratory differences while within-laboratory differences had only a marginal impact. The results can be used in the development of the standards.
... Method B in [17]. The DEDCs were truncated at the times corresponding to a level of 20 dB above the noise floor to compensate for errors introduced by limiting the integration interval [18]. The decay times T (θ q , φ q ) were estimated from the respective DEDCs using linear regression in accordance with the T 20 estimation specified in the international standard ISO-3382 [19]. ...
The absorption coefficient estimation in a reverberation room according to ISO 354 implicitly assumes a diffuse sound field and a uniform damping of the room modes composing it. This requirement is violated when an absorbing specimen is mounted inside the room, resulting in decay curves with multi-exponential slopes. Based on an analytic model, Hunt [Hunt et al., 1939] showed that the multi-exponential slope of the decay curve can be calculated as a superposition of groups of the axial, tangential and oblique modes of vibration and their corresponding damping caused by the finite impedance at the respective room boundaries.
Using array processing we decompose the sound field in a reverberation room into plane waves, expanding the wave number spectrum. Based on the wave number spectrum we apply a grouping of the different modes and investigate their respective damping using the directional energy decay curve calculated as the Schroeder integral for the respective time domain plane wave density function.
... Ma et al. [20] 160ms Peak selection and smoothing was performed, and the T 20 was calculated, based on work by [23], as ...
Dynamic range compression (DRC) is a very commonly used audio effect. One use of DRC is to emphasise transients in an audio signal. The aim of this paper is to present an approach for automatically setting dynamic range compression timing parameters, adaptively, allowing parameters to adapt to the incoming audio signal, with the aim of emphasising transients within percussive audio tracks. An implementation approach is presented.
... For recording the sweep responses, we used a 1 / 2 diffuse-field microphone (Type 4134, Brüel & Kjaer, Naerum, Denmark) with microphone preamplifier (Type 2669, Brüel & Kjaer, Naerum, Denmark) and microphone conditioner (Type 2690, Brüel & Kjaer, Naerum, Denmark). Room acoustic parameters T 30 and C 50 were calculated per source-receiver combination with MATLAB and the ITA-Toolbox [73], applying noise detection and compensation [78], with subsequent energetic averaging over spatial measurement results per octave band. ...
Virtual acoustic environments have demonstrated their versatility for conducting studies in various research areas as they allow easy manipulations of experimental test conditions or simulated acoustic scenes, while providing expansion possibilities to related interdisciplinary and multimodal fields. Although the evolution of auditory and cognitive models is consistently pursued, listening experiments are still considered the gold standard, usually necessitating a large amount of resources, including travel expenses of study participants. In order to facilitate practical and efficient study execution, we therefore implemented a mobile hearing laboratory by acoustically optimising the interior of a caravan. All necessary technical facilities were integrated to perform listening experiments in virtual acoustic environments under controlled conditions directly on site, for example, in front of schools or senior residential centers. The design and construction of this laboratory are presented and evaluated based on insulation properties, selected room acoustic parameters, and interior ambient noise levels that are to be expected during operation at representative test sites. Limitations, particularly in low-frequency insulation performance, should provide incentives for further optimisations in similar future projects.
... This operation is required by ISO 3382-2 standard to obtain reverberation time from an impulse response. The determination of the integration limit t can be especially problematic, and the work of Lundeby et al. [12] has been taken as the main reference to automatize this task. In particular, the background noise is calculated based on the final 10% of the decay curve, while the initial portion of the curve must be processed with a series of smoothing and piecewise segmentation operations; to estimate the decay slope, a dynamic range of 10 dB is normally evaluated starting 10 dB above the noise level. ...
The DENORMS Round Robin Test (RRT) is intended to study and improve the techniques used for the determination of the sound absorption coefficient of materials, with particular focus on the low-frequency range and on measurements in reverberation rooms. It is well known that the main reason why it is difficult to extend the frequency range of interest below 100 Hz is the low modal density. The modal behavior of a room is dependent on its geometry and this is one of the reasons why a discrepancy of the results coming from different laboratories can be found even if the same material is tested. This paper describes the measurement procedure developed to allow uniform acquisition and post-processing of acoustic response data of laboratories participating in the RRT, with and without absorbing materials inside. The tests and the post-processing operations performed on the measured data are also discussed in the paper.
... Reliability of the results remains important: the better the renderings, the more immersive the simulation, the more people engage themselves in the project, as reported by an interviewee. There are still known factors impacting the reliability of the results such as having proper anechoic material, partially linked to the precision of the associated acoustical measurements, dependent on the recording engineer/researcher, microphones quality, or signal-to-noise ratios (Lundeby et al., 1995;P€ atynen et al., 2008;Vorl€ ander, 2013). The variability of simulation algorithms may have an impact on the confidence acoustical consultants have in the results of auralizations, compared to traditional methods using objective parameters synthetically presented in written reports. ...
While auralization technology is used in a variety of fields, particularly in architectural acoustics, there is a lack of data on the auralization tools used and actual practices. In this perspective, this work presents the results of a survey study on auralization uses in the acoustical design and consulting community, targeting acoustical consultants. The objectives are (1) to identify the tools and methods used by acousticians to create auralizations as well as effective uses so as to understand the benefits and changes provided by this technology, and (2) to highlight the difficulties and limitations linked to the use of auralizations in concrete projects. Based on the theory of acceptability and use of technology, the study was conducted from a mix of quantitative and qualitative data collection approaches, combining a questionnaire answered by 74 respondents with semi-directed interviews with nine practitioners. Results highlight the main uses of auralizations, the diversity of projects in which auralizations are applied, and how auralizations are currently used in real-world situations. The benefits of using this technology, inherent weaknesses in the tools, and practical difficulties are also discussed.
... For recording the sweep responses, we used a 1 / 2 diffuse-field microphone (Type 4134, Brüel & Kjaer, Naerum, Denmark) with microphone preamplifier (Type 2669, Brüel & Kjaer, Naerum, Denmark) and microphone conditioner (Type 2690, Brüel & Kjaer, Naerum, Denmark). Room acoustic parameters T 30 and C 50 were calculated per source-receiver combination with MATLAB and the ITA-Toolbox [73], applying noise detection and compensation [78], with subsequent energetic averaging over spatial measurement results per octave band. ...
Research in the last decades has pointed out negative effects of noise on cognitive processing and general health. Depending on the frequency and level of exposure as well as individual shielding potential, risk groups like children or elderly people may be particularly vulnerable to commonly known consequences including a reduced attention span, influencing the learning success during education, increased blood pressure and cardiovascular problems or sleep disturbances. To better understand the susceptibility to noise, specifically designed listening experiments in controlled laboratory environments are indispensable but might put high demands on older study participants whose travel expenses to test locations could increase due to reduced mobility. For children, organizational issues like insured transport and supervision during waiting periods, necessitating additional resources, need to be resolved. In order to facilitate practical and efficient study execution, we therefore implemented a mobile hearing laboratory by acoustically optimizing the interior of a caravan. All necessary technical facilities were integrated to perform listening experiments in virtual acoustic environments under controlled conditions directly on site, for example, in front of schools or senior residential centers. The design and construction of this laboratory are presented here and evaluated through acoustic measurements.
... The measurement signal was amplified using a custom-made class B power amplifier and played back through the measurement LS, with two averages per source-receiver combination. Reverberation times T 30 were calculated using the ITA-Toolbox (Berzborn, Bomhardt, Klein, Richter, & Vorla¨nder, 2017), with applied noise detection and compensation according to Lundeby, Vigran, Bietz, and Vorla¨nder (1995). Mean results were obtained by arithmetically averaging respective parameter results per octave band. ...
Theory and implementation of acoustic virtual reality have matured and become a powerful tool for the simulation of entirely controllable virtual acoustic environments. Such virtual acoustic environments are relevant for various types of auditory experiments on subjects with normal hearing, facilitating flexible virtual scene generation and manipulation. When it comes to expanding the investigation group to subjects with hearing loss, choosing a reproduction system which offers a proper integration of hearing aids into the virtual acoustic scene is crucial. Current loudspeaker-based spatial audio reproduction systems rely on different techniques to synthesize a surrounding sound field, providing various possibilities for adaptation and extension to allow applications in the field of hearing aid-related research. Representing one option, the concept and implementation of an extended binaural real-time auralization system is presented here. This system is capable of generating complex virtual acoustic environments, including room acoustic simulations, which are reproduced as combined via loudspeakers and research hearing aids. An objective evaluation covers the investigation of different system components, a simulation benchmark analysis for assessing the processing performance, and end-to-end latency measurements.
The use of voice control of unmanned aerial vehicles is relevant due to the ease of practical use and new opportunities. This technology allows one to simplify the interface, making it more intuitive and natural. However, the quality and intelligibility of speech signals indoors can be significantly impaired by noise and reverberation. Therefore, before using voice technologies, it is desirable to take into account the effect of interferences by preliminary assessment of their parameters. In this paper, an algorithm for estimating the boundary (truncation time) between the informative and non-informative parts of the room impulse response, which allows obtaining believable estimates of the reverberation time, is proposed. The proposed algorithm is two-stage. At the first stage, “rouhg” envelope of the room impulse response is calculated using the detector-integrator, which allows one to find an approximate value of truncation time and obtain an approximate estimate of the reverberation time. In the second stage, output data of the first stage are used to refine the truncation time and reverberation time estimates. Experimental tests using recordings of real room impulse responses testify to the efficiency of the proposed algorithm.
REVERBDATA is a Room Impulse Response (RIR) database of significant Portuguese spaces, considering their acoustical, architectural, cultural, and functional significance. Multi-channel IR recordings are made at various points within these spaces, processed, and added to the database, accessible through a user-friendly interface for navigation and artificial reverberation generation. Utilizing advanced measurement and processing techniques, diverse impulse responses in different multi-channel formats are acquired to enable immersive sound reproduction via FFT convolution reverb. While existing RIR databases like Aachen Impulse Response (AIR), MOTUS, MIT IR Survey, Aalto Acoustics Laboratory, and OpenAIR offer extensive collections, they predominantly feature spaces that are common with limited representation of significant ones. The REVERBDATA Project develops a Portuguese acoustical heritage database characterizing significant architectonic spaces, thus serving as a valuable resource for professionals in sound engineering, architecture, and interior design, but also in the social sciences. It encompasses various spaces such as concert halls, theatres, churches, and other culturally significant venues nationwide. Detailed information including ISO 3382-1 acoustical parameters, architectural features, dimensions, materials, and other relevant factors influencing acoustics is provided. Currently, the REVERBDATA database contains approximately 375 high-resolution impulse responses in formats such as AB stereo, binaural, and 1st, 2nd, and 3rd order Ambisonics, covering over 140 locations across 15 significant spaces.
Porous acoustic absorbers have excellent properties in the low-frequency range when positioned in room edges, therefore they are a common method for reducing low-frequency reverberation. However, standard room acoustic simulation methods such as ray tracing and mirror sources are invalid for low frequencies in general which is a consequence of using geometrical methods, yielding a lack of simulation tools for these so-called edge absorbers. In this article, a validated finite element simulation model is presented, which is able to predict the effect of an edge absorber on the acoustic field. With this model, the interaction mechanisms between room and absorber can be studied by high-resolved acoustic field visualizations in both room and absorber. The finite element model is validated against transfer function data computed from impulse response measurements in a reverberation chamber in style of ISO 354. The absorber made of Basotect Ò is modeled using the Johnson-Champoux-Allard-Lafarge model, which is fitted to impedance tube measurements using the four-microphone transfer matrix method. It is shown that the finite element simulation model is able to predict the influence of different edge absorber configurations on the measured transfer functions to a high degree of accuracy. The evaluated third-octave band error exhibits deviations of 3.3-4.1 dB computed from third-octave band averaged spectra.
An established model for sound energy decay functions (EDFs) is the superposition of multiple exponentials and a noise term. This work proposes a neural-network-based approach for estimating the model parameters from EDFs. The network is trained on synthetic EDFs and evaluated on two large datasets of over 20 000 EDF measurements conducted in various acoustic environments. The evaluation shows that the proposed neural network architecture robustly estimates the model parameters from large datasets of measured EDFs while being lightweight and computationally efficient. An implementation of the proposed neural network is publicly available.
In the low frequency range, the reverberation of a room should be characterised by the decay time, which can be determined experimentally by various methods. In order to make accurate and precise measurements, the differences between these methods and their advantages should be known and quantified. In this work, the performance of four linear and four adaptive time-frequency impulse response analysis methods was evaluated with the aim to find the most accurate method for characterising the decays of the room modes. The methods were compared using generated Prony test signals with known modal properties and measured room impulse responses were also examined. The decay time using the selected time-frequency impulse response method was then estimated in three rooms and the measurement results were compared with interrupted sine tone methods. It has been shown that the choice of measurement setup and analysis parameters has an important influence on the estimation of the decay time and its modal frequency. The time-frequency impulse response method using the window width optimized Stockwell transform, the continuous wavelet transform, or the Morlet wave method were found to be the most robust methods, and their applicability is also enhanced by the fact that only a single measurement is required for the selected microphone position. The presented test methods could be used to evaluate the performance of other approaches for the evaluation of reverberation below the Schroeder frequency or the ability of measurement and signal processing techniques to characterize low frequency transient signals.
The exponential sine sweep is a commonly used excitation signal in acoustic measurements, which, however, is susceptible to non-stationary noise. This paper shows how to detect contaminated sweep signals and select clean ones based on a procedure called the rule of two, which analyzes repeated sweep measurements. A high correlation between a pair of signals indicates that they are devoid of non-stationary noise. The detection threshold for the correlation is determined based on the energy of background noise and time variance. Not being disturbed by non-stationary events, a median-based method is suggested for reliable background noise energy estimation. The proposed method is shown to detect reliably 95% of impulsive noises and 75% of dropouts in the synthesized sweeps. Tested on a large set of measurements and compared with a previous method, the proposed method is shown to be more robust in detecting various non-stationary disturbances, improving the detection rate by 30 percentage points. The rule-of-two procedure increases the robustness of practical acoustic and audio measurements.
O presente trabalho trata da medição de respostas impulsivas para o cálculo de dois parâmetros acústicos de salas: tempo de reverberação e fator de força. Todo o trabalho foi realizado com auxílio da linguagem de programação Python, com o objetivo de que as ferramentas desenvolvidas passem a integrar o pacote de códigos abertos Python in Technical Acoustics (PyTTa), dedicado a aquisição, processamento e visualização de dados em acústica e vibrações, idealizado e escrito por alunos do curso de graduação em Engenharia Acústica da Universidade Federal de Santa Maria (UFSM). O procedimento de Kirkeby para a regularização no cálculo de respostas impulsivas foi implementado. O algoritmo proposto por Lundeby foi utilizado na obtenção das curvas de decaimento energético, a partir das quais os resultados de tempo de reverberação foram obtidos. O cálculo do parâmetro é validado com a comparação entre os resultados obtidos em Python e os calculados em Matlab por meio do ITA toolbox. O cálculo do fator de força é extremamente dependente da potência da fonte sonora utilizada, e portanto um método de recalibração é empregado para corrigir eventuais variações na potência da fonte entre a sua calibração em câmara reverberante e a medição do parâmetro in situ. Um procedimento para validação dos cálculos do método da recalibração é proposto. Os resultados de fator de força são comparados com os obtidos de uma formulação estatística. A influência da não omnidirecionalidade da fonte sonora no valor dos parâmetros calculados é avaliada por meio de análise estatística. Os resultados do trabalho indicam algum grau de sucesso nas implementações realizadas. Este trabalho contempla a primeira abordagem do curso de engenharia acústica da UFSM na medição do fator de força de salas.
The paper deals with the influence of the pulse length on the decay of the sound field energy. Six pulse lengths— 2000, 2500, 3000, 3500, 4000 and 4500 ms—were selected for investigations.
Investigations show that a 2500 ms pulse is too short to correctly assess the background noise time interval. Such pulse length is not suitable for experiments. 3000 ms is the right length, while 3500 ms may be too long, resulting in errors of measurement results. When the pulse length increases to 4000 ms, the decay starting from 2000 ms is different from the pulse length 2500 ms and 3000 ms. Background noise starts from 2300 ms for these pulses, while for a 4000 ms pulse it starts from 3200 to 3300 ms. The length of 4500 ms is completely not suitable for investigations because the background noise zone starts very early, ie at 1800 ms, while for a short 2500 ms pulse it starts much later, after 2300 ms.
While investigating energy decay, it is important to determine the maximum decay. At 63 Hz the sound field decay is almost uniform till— 18 dB. Later the decay character is different. The decay of the longest (4500 ms) and the shortest (2500 ms) pulse after— 18 dB is very steep and reaches—30 dB. However, the decay is influenced by the background noise. Thus the shortest and the longest pulses are not suitable for the lowest frequencies. The greatest energy decay is characteristic of the 3000 ms pulse. After 1700 ms energy decreases to—30 dB. Thus at this frequency one may measure the echoing time while approximating decay from 0 to—20 dB.
As the frequency increases, the results change. At 100 Hz the energy decays by— 35–37 dB at pulse lengths of 2500 ms and 4000 ms. The greatest decay of— 42 dB is produced by the longest pulse 4500 ms though this arouses certain doubts. Then the echoing time may be measured from 0 to— 30 dB.
At 125 octave frequency the smallest maximum decay of— 40 dB is observed with the shortest pulse (2500 ms), while the largest one— 50 dB is produced by the longest pulse (4500 ms). Thus standard echoing time may be measured for this frequency.
In the frequency range of 250–2000 Hz, the maximum energy decay is sufficient and amounts to— 50–60 dB. At 4000 Hz the final part of decay is strongly dependent on the pulse length although, as the decay is about— 55 dB in all cases, the standard echoing time may be measured correctly.
Pulse length is important only for the calculation of the low-frequency echoing time. At 63–100 Hz the best maximum decay is seen with the pulse 3000 ms long, while at 125 Hz and over the best pulse lengths are from 3000 to 4000 ms. When the hall contains audience and tapestries are on the walls, the energy decay is almost uniform at the pulse lengths of 2000 to 2800 ms. In this case a better decay is obtained with the longest pulse of 2800 ms.
First Published Online: 26 Jul 2012
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