Active Sound Sources Research Articles

Theoretical and experimental study of transient sound field reconstruction in semi-free field

The interpolated time-domain equivalent source method (ITDESM) is based on the assumption of a free field, which might not be suitable for the direct application to a semi-free field. Here, an extended ITDESM procedure is proposed to reconstruct the semi-free transient sound field directly in the time domain. In this method, the time-domain equivalent sources are placed not only near the actual sound source, but also near its image source. It is assumed that the number of the equivalent sources distributed around the image source (virtual equivalent sources) is identical to that distributed around the actual source (actual equivalent sources), and their locations are mirror-symmetric with the symmetry plane being the reflecting surface. The acoustic contribution from each virtual equivalent source is further formulated with the strengths of the corresponding actual equivalent source to save the computational cost. Reconstruction of the semi-free transient sound fields from three monopoles is contrastively studied by numerical simulations to demonstrate the superiority of the proposed method. This method is also examined by adding noise and measurement point offsets into the reconstruction process, respectively. An experiment with a loudspeaker is conducted in a semi-anechoic chamber, and the results show the validity and applicability of the proposed method.

Reconstruction of semi-free transient sound field under different reflection conditions using an extended interpolated time-domain equivalent source method

Transient acoustic field can be rebuilt directly in the time-domain via the interpolated time-domain equivalent source method (ITDESM). However, this method requires that the reconstruction should be addressed in the free-field only, which can hardly be met in the engineering noise problems. To circumvent this difficulty, an extended ITDESM procedure is developed by extending the ITDESM from the free-field to the semi-free-field. In this approach, the time-domain equivalent sources are placed not only near the actual sound sources but also around their image sources with respect to the planar reflecting surface. The solving procedure of the equivalent source strengths is improved to decrease the computing load. The reflection conditions treated here can be arbitrary, i.e. both perfectly rigid and impedance-effected. Reconstruction results of the transient sound field radiated from three monopoles under different reflection conditions demonstrate the validity and applicability of the proposed method.

Classification of reverberant audio signals using clustered ad hoc distributed microphones

In a real world scenario, the automatic classification of audio signals constitutes a difficult problem. Often, reverberation and interfering sounds reduce the quality of a target source signal. This results in a mismatch between test and training data when a classifier is trained on clean and anechoic data. To classify disturbed signals more accurately we make use of the spatial distribution of microphones from ad hoc microphone arrays. In the proposed algorithm clusters of microphones that either are dominated by one of the sources in an acoustic scenario or contain mainly signal mixtures and reverberation are estimated in the audio feature domain. Information is shared within and in between these clusters to create one feature vector for each cluster to classify the source dominating this cluster. We evaluate the algorithm using simultaneously active sound sources and different ad hoc microphone arrays in simulated reverberant scenarios and multichannel recordings of an ad hoc microphone setup in a real environment. The cluster based classification accuracy is higher than the accuracy based on single microphone signals and allows for a robust classification of simultaneously active sources in reverberant environments.

Blind separation of sound sources from the principle of least spatial entropy

The aim of the paper is to offer a method for separating incoherent and compact sound sources which may overlap in both the space and frequency domains. This is found of interest in acoustical applications involving the identification and ranking of sound sources stemming from different physical origins. The principle proceeds in two steps, the first one being reminiscent to source reconstruction (e.g. as in near-field acoustical holography) and the second one to blind source separation. Specifically, the source mixture is first expanded into a linear combination of spatial basis functions whose coefficients are set by backpropagating the pressures measured by an array of microphones to the source domain. This leads to a formulation similar, but no identical, to blind source separation. In the second step, these coefficients are blindly separated into uncorrelated latent variables, assigned to incoherent “virtual sources”. These are shown to be defined up to an arbitrary rotation. A unique set of sound sources is finally recovered by searching for that rotation (by conjugate gradient descent in the Stiefel manifold of unitary matrices) which maximizes their spatial compactness, as measured either by their spatial variance or their spatial entropy. This results in the proposal of two separation criteria coined “least spatial variance” and “least spatial entropy”, respectively. The same concept of spatial entropy, which is central to the paper, is also exploited in defining a new criterion, the entropic L -curve, dedicated to determining the number of active sound sources. The idea consists in considering the number of sources that achieves the best compromise between a low spatial entropy (as expected from compact sources) and a low statistical entropy (as expected from a low residual error). The proposed methodology is validated on both laboratory experiments and numerical data, and illustrated on an industrial example concerned with the ranking of sound sources on a Diesel engine. At the same time, its robustness to the estimated number of active sources is demonstrated. • A method is proposed to blindly separate sound sources (i.e. without references). • Sources are assumed mutually incoherent and compact in space. • Two criteria, of least spatial entropy and of least spatial variance, are proposed. • The number of active sound sources is determined from an entropic L -curve. • The blind source separation method demonstrates good robustness.

Attention to memory: orienting attention to sound object representations

Despite a growing acceptance that attention and memory interact, and that attention can be focused on an active internal mental representation (i.e., reflective attention), there has been a paucity of work focusing on reflective attention to 'sound objects' (i.e., mental representations of actual sound sources in the environment). Further research on the dynamic interactions between auditory attention and memory, as well as its degree of neuroplasticity, is important for understanding how sound objects are represented, maintained, and accessed in the brain. This knowledge can then guide the development of training programs to help individuals with attention and memory problems. This review article focuses on attention to memory with an emphasis on behavioral and neuroimaging studies that have begun to explore the mechanisms that mediate reflective attentional orienting in vision and more recently, in audition. Reflective attention refers to situations in which attention is oriented toward internal representations rather than focused on external stimuli. We propose four general principles underlying attention to short-term memory. Furthermore, we suggest that mechanisms involved in orienting attention to visual object representations may also apply for orienting attention to sound object representations.

The use of interpolated time-domain equivalent source method for reconstruction of semi-free transient sound field

A semi-free transient sound field is reconstructed by extending the interpolated time-domain equivalent source method (ITDESM) in the free field to the semi-free field. In this approach, the time-domain equivalent sources are placed not only near the actual sound sources, but also in the vicinity of their mirrored sources with respect to the reflecting plane surface. Suppose that the number of equivalent sources distributed around the mirrored sources (virtual equivalent sources) is the same as that used near the actual sources (actual equivalent sources), with their locations symmetrical about the reflecting surface. The reflecting surface considered here can be perfectly rigid as well as impedance-effected. Furthermore, by reformulating the strengths of the virtual equivalent sources at each time instant with those of the corresponding actual equivalent sources, the computation load of solving the equivalent source strengths can be reduced by 50%. Numerical examples of reconstructing the semi-free transient sound field radiated from three monopoles under different reflection conditions demonstrate the feasibility of the proposed method.

Comparison of precedence effect behavior in anechoic chamber with that in ordinary room

The precedence effect is well known as one of auditory illusions occurred by using multiple sound sources with similar sound output. When a sound is followed by similar sound separated with relatively short time delay, a single fused sound image is localized at the source position corresponding to the first-arriving sound. This feature is applicable to public address systems, which make audience perceive the sound image different from the actual sound source positions prepared for the system, with some sound reinforcement achieved. In spite of many studies in this phenomenon, the behavior of the precedence effect has been investigated for limited sound source arrangements in laboratory environments like anechoic chamber. On the other hand, this behavior in the ordinary room is not obvious, and it is effective to clarify the difference of the behavior of the precedence effect in anechoic chamber from that in the ordinary room for the application of the precedence effect to the public address system. In this study, the similar sound sources were installed both in the lecture room and in the anechoic chamber, and the behavior of the precedence effect was compared each other with the given time and level difference among sound sources.

Effect of distant-variant/invariant head-related transfer functions on perception of a proximal sound source in virtual auditory space

A virtual auditory space can be presented to a listener based on binaural synthesis using head-related transfer functions (HRTFs) that are obtainable by measurements or numerical simulations. Due to hardware complexity, HRTF measurements are typically made for a fixed source distance though they are used in binaural synthesis for variable source distances. However, it is known that HRTFs depend on source distance especially for proximal sources for distance less than 1 m. So it is possible that binaural synthesis with HRTFs for a fixed source distance may result in degradations for proximal sound image perception. In this paper, experiments were performed to examine how the use of distant-variant or -invariant HRTFs affect the perception of a proximal sound source in a virtual auditory space in which the listener’s motion is compensated by head tracking. HRTFs for source distances up to 1 m, in 5 cm steps, are numerically simulated using the boundary element method. Results show the difference between presented and perceived source distances being significantly smaller when distance-variant HRTFs were used. This indicates that the use of HRTFs corresponding to actual sound source position leads to accurate perception of a proximal source.

Method for the Separation of Vehicle Interior Noise Contributions Using Only Response Signals

In this study, we applied an independent component analysis (ICA) to separate the contributions of vehicle interior noise by using only response signals. To verify the applicability of ICA, we utilized engine and wind noise and integrated them into a simple simulation with three conditions (synchronous, asynchronous, and asynchronous with frequency characteristic), and identified their separate contributions by applying ICA to these mixed response signals. Results show contributions were calculated accurately while using time-domain ICA for the case in which source signals were synchronously mixed, and it was found that accuracy was low when the signals were asynchronously mixed. Frequency-domain ICA was found to separate the contributions correctly. When the source signals were asynchronously mixed with specific frequency characteristic, permutation problems occurred, in which the matching between calculated independent component by frequency-domain ICA and the actual sound source changed among frequency. Therefore, a permutation solution method using a non-Gaussian characteristic of the calculated contributions was proposed. As a result, the contributions could be calculated accurately.

Capturing and Reproducing Spatial Audio Based on a Circular Microphone Array

This paper proposes a real-time method for capturing and reproducing spatial audio based on a circular microphone array. Following a different approach than other recently proposed array-based methods for spatial audio, the proposed method estimates the directions of arrival of the active sound sources on a per time-frame basis and performs source separation with a fixed superdirective beamformer, which results in more accurate modelling and reproduction of the recorded acoustic environment. The separated source signals are downmixed into one monophonic audio signal, which, along with side information, is transmitted to the reproduction side. Reproduction is possible using either headphones or an arbitrary loudspeaker configuration. The method is compared with other recently proposed array-based spatial audio methods through a series of listening tests for both simulated and real microphone array recordings. Reproduction using both loudspeakers and headphones is considered in the listening tests. As the results indicate, the proposed method achieves excellent spatialization and sound quality.

A Bayesian inference model for speech localization (L)

The localization of active speakers with microphone arrays is an active research line with a considerable interest in many acoustic areas. Many algorithms for source localization are based on the computation of the Generalized Cross-Correlation function between microphone pairs employing phase transform weighting. Unfortunately, the performance of these methods is severely reduced when wall reflections and multiple sound sources are present in the acoustic environment. As a result, estimating the number of active sound sources and their actual directions becomes a challenging task. To effectively tackle this problem, a Bayesian inference framework is proposed. Based on a nested sampling algorithm, a mixture model and its parameters are estimated, indicating both the number of sources-model selection-and their angle of arrival-parameter estimation, respectively. A set of measured data demonstrates the accuracy of the proposed model.

G100044 等価的音源同定に基づく音場推定法開発のための基礎研究

Under severe acoustic environments such as rocket launching sites, we need to pay particular attention to sound pressure field. However, no numerical method is currently available enough for accurate mapping of sound pressure level especially in case of large sound field due to unclear location and distribution of sound sources. It is then desired to develop new methods to realize accurate estimation of sound pressure field. In this paper the authors propose a new estimation method for the technical desire. The method is basically built up based on the boundary element method and a new equivalent sound source identification for modeling actual sound sources. Furthermore, since the number of the velocity parameters to be determined is more than that of sound pressure measurement points in practice, the singular value decomposition method is used in the solution process of the inverse problem in the identification. Experiment is also carried out for verification using sound of 300Hz and 600Hz in a semi-anechoic camber. As a result, it is proved that the estimation method has enough potential ability to estimate acoustic fields with a practical accuracy.

TDOA Estimation for Multiple Sound Sources in Noisy and Reverberant Environments Using Broadband Independent Component Analysis

In this paper, we show that minimization of the statistical dependence using broadband independent component analysis (ICA) can be successfully exploited for acoustic source localization. As the ICA signal model inherently accounts for the presence of several sources and multiple sound propagation paths, the ICA criterion offers a theoretically more rigorous framework than conventional techniques based on an idealized single-path and single-source signal model. This leads to algorithms which outperform other localization methods, especially in the presence of multiple simultaneously active sound sources and under adverse conditions, notably in reverberant environments. Three methods are investigated to extract the time difference of arrival (TDOA) information contained in the filters of a two-channel broadband ICA scheme. While for the first, the blind system identification (BSI) approach, the number of sources should be restricted to the number of sensors, the other methods, the averaged directivity pattern (ADP) and composite mapped filter (CMF) approaches can be used even when the number of sources exceeds the number of sensors. To allow fast tracking of moving sources, the ICA algorithm operates in block-wise batch mode, with a proportionate weighting of the natural gradient to speed up the convergence of the algorithm. The TDOA estimation accuracy of the proposed schemes is assessed in highly noisy and reverberant environments for two, three, and four stationary noise sources with speech-weighted spectral envelopes as well as for moving real speech sources.

Improving Power Spectra Estimation in 2-Dimensional Areas Using Number of Active Sound Sources

An improvement of estimating sound power spectra located in a particular 2-dimensional area is proposed. We previously proposed a conventional method that estimates sound power spectra using multiple fixed beamformings in order to emphasize speech located in a particular 2-dimensional area. However, the method has one drawback that the number of areas where the active sound sources are located must be restricted. This restriction makes the method less effective when many noise source located in different areas are simultaneously active. In this paper, we reveal the cause of this restriction and determine the maximum number of areas for which the method is able to simultaneously estimate sound power spectra. Then we also introduce a procedure for investigating areas that include active sound sources to reduce the number of unknown power spectra to be estimated. The effectiveness of the proposed method is examined by experimental evaluation applied to sounds recorded in a practical environment.

Design of critical listening rooms: A historical perspective.

Critical listening room design has progressed to follow loudspeaker evolution from mono to two-channel to multi-channel, as well as the advances in loudspeaker quality. There are basically two fundamental approaches to two-channel room design. The non-environment room, in which the contribution of the room is minimized, and versions of live-end-dead-end, in which early reflections were minimized creating a spatio-temporal reflection free zone and diffuse reflections from the rear of the room are used to create passive surround sound. This presentation will focus on an immersive design for multi-channel loudspeaker formats, which utilizes broad bandwidth absorption and diffusion. Research with low-frequency plate resonators and a broad bandwidth diffusion chamber has contributed to the proposed design and results will be discussed. The proposed multi-channel critical listening room includes the use of low-frequency control down to 40 Hz provided by 4-in.-thick metal plate resonators on a portion of the front wall and in all corners, broad bandwidth diffusion on all wall and ceiling surfaces, and strategically placed multiple subwoofers. The intent is to minimize monophonic reflections and create lateral enveloping energy by uniformly scattering sound from all of the active sound sources in the room.

High-Resolution Imaging of Sound Sources in Free Field Using a Numerical Time-Reversal Sink

Numerous practical applications ― such as non destructive evaluation of industrial structures, acoustic characterization of musical instruments, and acoustic mapping of sound sources in a known propagation medium ― involve source detection and characterization. In the past, this problem has been investigated using different beamforming and backpropagation methods. In this work, a new technique, based on the time reversal sink concept, is used to detect active sound sources with a limited number of measurement points. The theory and application of super-resolution focusing of sound and vibration using a time-reversal sink (TRS) have been studied, both in ultrasonic regime and in audible range. A high-resolution imaging technique based on a numerical time reversal sink has recently been developed by the authors for vibrational imaging of active sources in a dispersive medium. In this paper, the numerical time reversal sink imaging technique is adapted to the case of high-resolution acoustic imaging of active sound sources in a three-dimensional free field. This technique allows high-resolution imaging and provides a new method of characterization and detection of sound sources. All results show the high resolution imaging capabilities of this new technique when compared with classical time-reversal (TR) backpropagation. More than simply detecting the position of the acoustic source, this technique allows to detect the size of the active sources. This technique provides an alternative to other imaging and source detection techniques, such as three-dimensional acoustic holography and beamforming.

Comparison of Sound Pressure Distribution Determined by Numerical Analysis and Scaled-Up Experiment for Small Ultrasonic Probe with Lens

Ultrasonic medical equipment is used in not only diagnosis but also therapy such as that for treating hyperthermia. Many researchers are also studying high-frequency ultrasonic imaging systems with interluminal or catheter transducers. In both applications, an acoustic lens might improve the characteristics of ultrasonic medical probes. In this paper, a small acoustic lens for an ultrasonic catheter-type probe is described. The conventional three-dimensional finite-difference time-domain (3D FDTD) with orthogonal coordinates requires a large memory and a long calculation time to estimate the characteristics of the lens. To overcome these disadvantages, a simple two-dimensional (2D) FDTD calculation based on symmetry is proposed in this paper. A virtual spherical sound source whose amplitude distribution is equal to that of the sound propagation field of an actual sound source is also used to simplify the calculation. A numerical model of the lens with a lens holder is constructed. The experimental results agree well with the calculated results with the lens holder.

무향실에 있어서의 음압레벨과 거리정위와의 관계

According to a lot of investigations, distance perception is influenced by many important cues such as sound pressure level, reflections from the room surface, binaural difference (ITD and ILD), a kind of sound source, and head related transfer functions (HRTF). Two psychoacoustical experiments on auditory distance perception were conducted to examine the effectiveness of the sound pressure level loudness as one of the physical cues in the auditory distance perception under a constant loudspeaker`s output level and a constant sound level at the subject`s position in the absence of reflections in an anechoic room. Our experimental results showed that the perceived distance of sound image is closer than actual sound source distance with the constant loudspeaker`s output level and the constant sound level. Futhermore, the perceived distance of a sound image with constant sound level increased when the actual distance increases up to approximately 2 m while the perceived distance saturated when the sound source distance exceed 2 m. On the other hand, when the condition of loudspeaker`s output level was kept constantly, the perceived distance of sound image increased up to around 3m, longer than the conditions of constant sound level at the subject`s position. We found that the change in the loudness as a function of distance plays an important role in the auditory distance perception in the absence of reflections..

Super-resolution imaging of active sound and vibrational sources using a time-reversal sink

Theory and experiments of super-resolution focusing using a time-reversal sink have been investigated in high-frequency regime [Rosny and Fink, Phys. Rev. Lett. 89] and in audible range [Bavu, Besnainou, Gibiat, Rosny and Fink, Act. Acoust., 93]. This technique, generalized to the case acoustic and vibrational imaging of active sources, allows super-resolution imaging and provides a new method of characterization of active sources in a known background medium. This imaging technique involves a measurement in the background medium using an array, and the simulation of the backpropagating-field in a fictive medium. An ideal numerical time-reversal sink (NumTRAS) is then used to refine results and obtain high-contrast, high-resolution imaging of initial sources. The algorithm has been validated in parallel supercomputer simulations, in both vibrational and acoustics fields and has been used to detect active vibrational sources in a clamped Mindlin plate and active sound sources in an anechoic room. All results show high-resolution imaging capabilities when compared with classical time-reversal backpropagation. NumTRAS provides an alternative to other imaging and source detection techniques, such as acoustic holography and beamforming. Beyond the applications of acoustic and vibrational non-destructive evaluation of industrial structures, NumTRAS has applications in evaluation of musical structures and is being tested to detect and characterize moving sources.

Robust tracking of moving sound source using scaled unscented particle filter

Tracking an active sound source involves the modeling of non-linear non-Gaussian systems. To address this problem, this paper proposed scaled unscented particle filter (SUPF) algorithm for tracking moving sound source. The particle filter part of the SUPF provides the general probabilistic framework to handle non-linear non-Gaussian systems, and the scaled unscented Kalman filter (SUKF) part of the SUPF generates better proposal distributions by taking into account the most recent observation. Meanwhile, models used in SUPF algorithm were also explored for the sound source motion, observation and the likelihood of the sound source location in the light of the Langevin process. Compared with the conventional PF approach, the simulated results demonstrated that the SUPF algorithm had superior tracking performance.